CLTECH: Cisco Collaboration Infrastructure Overview

0:00[AUDIO LOGO] So in this skill, we go over what exactly is a Cisco Certified Technician and what it exactly means to be CCT Collaboration certified. We also go over the infrastructure overview in regards to a collaboration environment. We talk about voice gateways, dial peers, and call legs. Then finally, we discuss Cisco converged networks. So with that, I thank you for taking the journey with me, and I'll see you in the next video.

0:00[AUDIO LOGO] So what exactly is a Cisco Certified Technician, or a CCT Certification? That's exactly what we'll be talking about in this Nugget. We'll go over what it means to be a Cisco Certified Technician. We'll go over the CCT Collaboration and the topics in that exam. And we'll also discuss how the CCT compares to other physical certifications, like CCNA and CCNP. Let's get started and dive in.

0:30When it comes to a Cisco Certified Technician, these are entry exams that one can take, especially when they're new to the industry. So you may ask, I've heard of CCNA, or I've heard of CCNP, but what is a Cisco Certified Technician? So let's go over to Cisco's site. And we see here that we have a CCT, a Cisco Certified Technician for Collaboration, for Data Center, and also for Routing and Switching. So as we see, this is an entry-level certification. And especially if you're new to collaboration, or data center, or routing and switching, these certifications can help you get familiar with the technology to help you support them.

1:09You see the CCT Collaboration really focuses on the skills for on-site support and the maintenance of Cisco collaboration endpoints and operating systems. So you may be the front line in support. So when there's an endpoint not working, it may be your job to further investigate and troubleshoot the issue. Or maybe you have to replace an endpoint that has gone bad. Or maybe you deploy these endpoints when a new site is spun up. But you become a critical piece of the puzzle because in the event something is not working, your other team members will look at you to see what troubleshooting steps that you perform to determine that we have an issue.

1:46So you would be expected to know the voice and video endpoints, the interfaces on these endpoints, along with the accessories. We also should be comfortable with the command-line interface, along with basic troubleshooting skills. And I'm running out of space there. But you may determine that some hardware and endpoint had to be replaced. Or if the upgrade on the software or firmware on the endpoint fails, you could troubleshoot as to why it failed. So looking at the Cisco certifications, we see Associate.

2:18We see Professional, and then we see the Expert. This chart doesn't have the entry certifications listed. So many folks with some experience already, they go toward their CCNA. Then depending on what track they want to go to, they go for their CCNP, the professional level. And then you have the option, if we choose to take that mission, to try for the CCIE. And that will be your expert level. And here at CBT Nuggets, we have some great training for many of these exams. So I personally worked on some of this content here at CBT Nuggets that help you get your CCNP Collaboration.

2:54But just know that the CCT Collaboration exam, again being entry, will prepare you for future tracks in your certification journey. But if certification is not your goal, it will definitely help you to be able to provide support for your environment when it comes to the endpoints for the collaboration environment. I want to note, if you go to this link here, right there, here we see additional information about certifications, including your entry-level certifications. And here's where you would find your entry-level certification of CCT.

3:25Then eventually throughout your career, you advance, and you can get higher tier certifications. So at this point, you may be thinking, what are some topics that I expect to see on the exam or be expected to be familiar with? So if you scroll down a bit, here are the topics that this particular exam will go over. We have the collaboration environment. That's 35%. And we can see this is an entry exam because notice a lot of the keywords-- describe, introduce, identify. And you may see for the CCNA or the CCNP exams, they use keywords such as explain, or troubleshoot, or configure, and deploy.

4:04So we have one section on the collaboration environment. We have another section just dedicated to different endpoints and hardware that you may see in the field. And then finally, we see a section here for service knowledge. This is a big chunk right there. So we verify-- or how do we verify IP connectivity? We also go over the command line and go over some commands. How do we manage the software on these endpoints? We also go over how we replace hardware due to a hardware failure. Maybe you're upgrading the endpoint. So even though this is an entry certification, there is a lot of good information here, especially if you're new to Cisco Collaboration.

4:40So in this Nugget, we went over what is a Cisco Certified Technician. We focus on the CCT Collaboration and also saw the topics for that exam. And also, we discussed the CCT compared to other physical exams or certifications, like the CCNA and the CCNP. So with that, I hope this has been informative for you. I'd like to thank you for viewing.

0:10Depending on how old or young you are, you may or may not remember having an analog phone line in your home. I remember my mother having a phone book with everyone's name and phone number notated that she would reference every time she would make a phone call. In this Nugget, we'll talk about the traditional voice system along with the voice over IP infrastructure, and we'll see what the benefits and differences are. So let's dive into our infrastructure overview. For many of us, when we were younger, we had an actual telephone within our home. We had a phone number assigned to that telephone line. And this was all possible because we had a voice circuit delivered from the PSTN.

0:44And this voice circuit was delivered via a copper wire. And this came from the CO, or the central office. You can think of the CO as this building that had a bunch of switching equipment inside that building. And the equipment belonged to the telephone company. And it's connection from the CO that was delivered to your home or to your business is what connected you to the PSTN. So then, within your home, you had the RJ11 port that this phone plugged into, and then once you went off hook, you heard a dial tone.

1:18So then, Pedro here may dial 773-555-7000. Then, if this pet store picks up, we now have a dedicated connection from Pedro's home to that pet store. And this was referred to as circuit switching. Why? Because we have a dedicated circuit established between both parties so they can talk to each other. So in this scenario, we had a dedicated phone number, and anyone can dial this phone number and make this phone inside Pedro's home ring. So when Pedro talks into his receiver there, his voice, or the sound wave of his voice, is converted to electrical signals.

1:59And the signals travel over that copper wire until they reach their destination. And then whoever Pedro is talking to over here, these signals then get converted back to some waves, and that's what that person hears. So with the analog line, we're using the properties of electricity to transmit your voice from one side to the other. So as I mentioned here, your phone was connected to an RJ11 port. So going out to the web, on Amazon, now it's kind of hard if you quickly glance at it, but you can see the differences between ports.

2:34This port is smaller in size than a network connection, which is an RJ45. Then we're looking at [INAUDIBLE] out here. Here's our RJ11 telephone plug. And then we see two pairs of copper wire. And here we have one, two, three, four, five, and six. So we have two pairs of copper wire here, but only three and four are used for the telephone line. Now the other pair, two and five, could be used for an additional line-- the takeaway being that we are only using one pair of copper wire for the phone connection.

3:10So with Pedro's example, we had one telephone at his home, but you easily could have a business that had multiple lines going to the building. So, for example, this building may have three sets of phone lines for each business. Larger companies had a PBX, but you could have lines connected to that PBX, and all your phones could be connected to that PBX, and that PBX may give you ability to do functions like call transfer, maybe voicemail, put a call on hold, but many features that you would find in an enterprise environment.

3:43But if calls were meant to be internal, then all that could be handled within the PBX itself. But in the event we need to make a phone call to the outside world, then the PBX would run that call for you. So as we stated, when Pedro here talks into the receiver, the sound of his voice gets converted to electrical signals. These signals is what's actually routed or sent over the infrastructure to the other side. And then those signals are converted back into sound waves. So the sounds coming out of Pedro's mouth is converted into electricity.

4:18And it works very well, but you could get light noise. So you may have static underlying could have a crackling sound going on. Maybe you have a buzzing, or some describe it as a hum. And if you've worked with analog lines before, you probably have seen all this. So what causes that line noise? Maybe you have bad wiring. And these could be lines that are not terminated properly or punched down, or maybe you could have rodents. These rodents are actually chewing up your infrastructure.

4:48And throughout my career, when we used to deal with analog lines and punch blocks, I've been to locations where wires have been chewed by rats. So for whatever reason, the rodents decided to chew through this wire, therefore, put in their bellies, or causing an interruption of service with that phone line at that location. We could have bad weather causing line noise. Remember, these copper wires are buried in the ground. So I remember one site that we had, every time it rained, it was guaranteed that we would get a phone call about that site having line noise.

5:20And come to find out, the ground around this building, when it would rain, the water would just collect. It looked like a lake. And what would happen is once that moisture would dry up, and then the line noise would go away. So rain or even lightning may be the issue there. You also may have an issue with corrosion. These analog lines may have been around for a very long time, so they may be exposed to chemicals or other things that may corrode the line and, therefore, inducing line noise.

5:50Then, finally, we have electrical interference, and that comes into play when you are close to a power line, for example. So the signals, sometimes, to reach a destination have to travel over long distances. So throughout the infrastructure, there could be many-- in fact, there is-- there's many central offices that a phone call may have to travel through. So to reach your destination, your call may go through this CO there and eventually be connected to this CO and then be routed to your destination.

6:23The takeaway here really being that these signals may have to travel a long distance. So there could be repeaters along the way, and the job would be to regenerate those signals along the way until they reach their destination. So here we are boosting the signal as it fades over long distances. But keep in mind, not only are we we're generating those electrical signals, which later are converted to our voice, but we're also regenerating any noise that may be online.

6:53So these repeaters not only amplified or regenerated Pedro's voice, but also the noise, therefore making that conversation between Pedro and the destination not too pleasant. But then we see this transition of going from analog to digital. And here, we are converting the sound of the voice to 1's and 0's. This is done using PCM, pulse-code modulation. And this is used to digitally represent the sampled analog signals. So PCM is converting the analog over to the digital.

7:28We're converting our voice through a series of numbers. Then, when we reach our destination, those series of numbers are converted back to our voice. So then this helps us with our line noise because the DigitalOcean voice stream is less susceptible to noise than analog signals, or those electrical signals being sent over that copper wire. You may have heard me use a key word when converting from analog to digital. We do so by sampling the analog sound. So the analog sound is sampled 8,000 times per second.

8:01So that's 8,000 samples, and each sample is assigned a numerical value. So we take the sound of your voice here, and this may be the sound wave, again, over 1 second. And then this is sampled 8,000 times. And then each one of these samples will have some type of numerical value to reference that sample. So human speech can be heard from 200 Hertz all the way to 9,000 Hertz. But you can understand human speech from 300 Hertz to about 3,400.

8:35And when I use the word understand, there's enough quality there in this range to really understand the speech from the other side, but also sense their mood. There was a gentleman named Harry Nyquist. I definitely suggest you look them up if you want this information in much greater detail. But he found that you can reconstruct these sound waves by taking samples that are twice the highest audio frequency being used. So to break that down, because it does sound confusing, he was looking at the range between 300 Hertz and 4,000 Hertz.

9:09So Nyquist wanted to reconstruct the audio stream by taking samples twice the highest audio frequency be used. So we have 4,000 times 2, which gives us our 8,000. And we're doing this every second. So we're taking 8,000 samples every second, and each sample is 8 bits, so if we say 8,000 times 8, that gives us our 64,000 bits per second. If this number looks familiar, each voice call is about 64 kilobits per second.

9:43This is also referred to as a DS0, and if you have worked with T1 and E1 circuits, each channel within that circuit is 64 kilobits per se. But the takeaway here is that we are taking our voice and really digitizing that voice by using numbers to represent the sound of our voice instead of the electrical signals, like with the analog circuit. Then we have voice over IP. For voice over IP, we're taking our voice, you're still converting it to numerical values, but this time, we're putting those values in a packet, and then sending that packet across the network.

10:18So our voice is being carried over the IP network. So we are not using telephone lines to send our voice. We're sending our voice over the internet. With voice over IP, we're using the internet to make and receive phone calls. So with voice over IP, we are reducing the cost of communicating because we don't need many phone lines or circuits going to our infrastructure because we are using the IP network. And because of that, we save tons on cabling because we can already use our wired or wireless infrastructure within our environment to carry our voice.

10:53So we have our savings here because we're going over the internet to make our phone calls. We save on cabling. And then we can use voice over IP via other applications within our infrastructure-- maybe voicemail or maybe teleconferencing or even instant messaging. So with voice over IP, we have many pieces that make up this puzzle. We have our IP phones. We also have our voice gateways that give us connectivity out to the PSTN. We could have other gateways like CUBE, Cisco Unified Border Element, and this connects us to an ITSP.

11:25Now what's the difference between ITSP and PSTN? PSTN, which is Public Switch Telephone Network-- with PSTN, we are connecting to those analog type of circuits that we've discussed so far. These are dedicated circuits that are established from our source to our destination. You may have a connection out to the ITSP, or the Internet Telephony Service Provider. This is where we route our calls or make or receive calls via the internet. We also have call agents, and our example here is CUCM, or Cisco Unified Communications Manager.

11:59And this is how we control our IP phones and the different endpoints in our environment. We have other application servers in our environment. You could have Cisco Unity Connection for voicemail. Or maybe we have a call center, so we have Contact Center Express or Contact Center Enterprise. You may have IM and Presence for messaging, or you could have a Cisco Meeting Server. The takeaway being that we could have other applications within our voice over IP infrastructure. You could have endpoints here that allow you to take part of a video conference.

12:30So this unit may have a camera and a microphone that you can use. So now you're using video and audio from the same endpoint. Now here, I've shown physical endpoints, but you could have software endpoints like Cisco Jabber. And Cisco Jabber [INAUDIBLE] cell phones allow you to use them just like you would use your endpoints to make or receive calls. And this could be on your desktop or laptop, maybe a tablet, or maybe your mobile device. So in voice over IP, we have some components that are needed.

13:02For example, we need signaling. And signaling helps us control, establish, and also terminate voice calls within our environment. So here we are exchanging information between two endpoints. And these messages, as we said, are used to set up and terminate the call. Now the PSTN uses a protocol called SS7, or Signaling System 7. And this protocol really defines how the PSTN exchanged these messages and control service.

13:32And the protocol does do more, like call forwarding, voicemail, call waiting, but SS7 is being used by the PSTN to set up the calls, route the calls, and turn them down. For voice over IP, we have multiple protocols that can be used, for example H. 323 and SIP, SIP standing for Session Initiation Protocol. But just like SS7, these are signaling protocols being used for voice over IP. So these are also used to set up and terminate and route the phone calls. Now H. 323 still being used today.

14:04It was once widely used, but now that title goes to SIP. But both of these are known as peer-to-peer protocols, which gave way to the latest CUBE, for example, the Cisco Unified Border Element, if running SIP, this has enough intelligence to set up and terminate the call itself-- those calls being made to the ITSP or those calls coming in from the ITSP. Now take the example of using other similar protocols, like the Skinny Protocol, or the session-- or Skinny Client Control Protocol, or MGCP, the Media Gateway Control Protocol.

14:43These are known as client-server protocol. So if you have a voice gateway running any of these two protocols here, for example, it's not intelligent enough to set up and terminate calls or even interpret call control messages. It needs a call agent like CUCM for example. And that call agent will manage that voice gateway and also provide call control. So let's say we had a analog telephone connected to this voice gateway. And someone picked up the receiver, and the phone went off hook.

15:14If voice gateway is running MGCP or the Skinny Protocol, does not know enough to send out on the phone. It needs to contact a call agent, being CUCM, and CUCM in return will tell that voice gateway to play dial tone. The real advantage here of using these client-server protocols is that the CUCM or the call agent in this example really manages those gateways down to the call control. So, for example, CUCM may have the dial plan, so the voice gateway does not have to be made aware of a dial plan.

15:47But in the event the CUCM is down, for whatever reason, then that voice gateway is not intelligent enough to interpret those call control messages. Let me make some room to screen here, but if we're running SIP, this gateway has enough intelligence to initiate and terminate those phone calls without the need of CUCM. It does require more configuration because you do have to configure a dial plan on that endpoint. That is not the case if you're using MGCP because you're are relying on that call agent to provide that dial plan for you.

16:19Now, in some networks, you might find a gateway that's called a gatekeeper. And in the event you had many voice gateways in your environment, the gatekeeper can help manage call routing between all of your gateways. The gatekeeper runs H. 323, but it provides Call Admission Control, which helps prevent the oversubscription of voice over IP networks. It also allows us to keep a certain level of audio quality within our network by providing bandwidth control and management. Now with SIP being so dominant, you may not see a gatekeeper much in many environments, but they're still out there.

16:52Then we have codecs and codec is really compression and decompression. And codecs are used to convert the analog signal over to a digital. So for the PSTN, G. 711 is the codec that is being used. G. 711 is being used to convert the analog to a digital 64 kilabit per second voice string. That's for the PSTN. When it comes to VoIP, you could use other codecs in your environment, for example G. 729.

17:27This also takes the analog signal and converts it to digital, but it uses 8 kilobits per second. And it does this using compression. In the event that bandwidth is an issue, that 8 kilobits per second looks more attractive to you on a protocol basis than 64 kilabits per second. So in the case of G. 729, not only are we performing this analog to digital conversion, but we're also adding compression. So when this package reaches destination, it then has to decompress those packets and then convert it from digital to analog.

18:01In this Nugget, we started our discussion about the infrastructure overview. We looked at the traditional telephone system and also discussed voice over IP. In the next Nugget, we'll continue our conversation by looking at voice gateways and see what part they play within our infrastructure. So with that, I hope this has been informative for you. I'd like to thank you for being.

0:10So, in this Nugget, we'll continue our infrastructure overview by talking about voice gateways. Voice gateways play a critical part in our environment by routing calls into and outside of our environment. We also talk about dial peers and call legs and the relationship between both. So let's get started. Start talking about voice gateways. Talking about voice gateways, they help CUCM, our Unified Communications Manager, to be able to communicate with non-IP devices these non-IP devices are really telephony devices. Maybe we're talking about an analog phone line, maybe we're talking about a PBX.

0:45So, those voice gateways are critical because they allow us to connect the PSTN to our voice Over IP infrastructure. Let's go on to Cisco's website and let's look at some of these gateways that are currently in production today. We have the Cisco 1900 Series. Now, we can see dates here as far as end-of-sale date, which was 2020. And then, end-of-support date, 2025. There are branch routers that can be used as voice gateways with the ability to insert certain modules in the back device, based on wherever your requirements are in regard to connecting to maybe a PBX or out to the PSTN.

1:21Those who have the 2800s Series, along with the 3900 Series, then followed by the 4000 Series. Now, at the time of this recording, this is probably the model that you will see deployed most, but it's always critical to always know when your products are either end-of-sale or end-of-support. So, we notice, going back to the 2900 Series routers, we see that the end-of-sale is 2017 and end-of-support is 2022. We don't see that for the 4000 Series. So, maybe you're expanding voice over IP to a few new sites and you want to purchase and deploy this model.

1:56Usually, you have to purchase this equipment through a Cisco partner and they should be recommending what hardware you should be purchasing. Let's take a look at documentation. And, if I do a search for sale, here we see end-of-sale and end-of-life announcement for select ISR products. So let's click on this. Now, the keyword here being "select." For the 4400 Series, there are certain models. You have a 4431. You have a 4451. And you also have a 4461. The 4431 and 51 are affected by this announcement, but the 4461 is not.

2:31So it's really important to know what's deployed in your environment when you're reading these announcements. So if I scroll down, we have an end-of-sale of November 2022. We're currently in July 2022, at the time of this recording. And then, the end-of-support date, we see, is 2027. So, we have these still deployed and they're under contract, we should be good. But, in regards to purchasing, we or your Cisco partners should be doing some research as far as what we should be purchasing long-term. And here, we have the Cisco Catalyst 8300 Series.

3:01And, if you scroll down a bit, here's a picture of them. We see some information about this platform, but I want to scroll down. We have a section of Unified Communications, so here, we see we have some voice functions like the SRST and that stands for Survivable Remote Site Telephony. In the event that WAN connection is down, you still have some limited functionality when it comes to your telephony. And that's critical for sites that really depend on that function. And we also see that this can function as a CUBE. And this can run a version of Communication Manager, that being, Communication Manager Express.

3:34So my takeaway here is that there's many different options that we can deploy when it comes to our voice gateways, but we always want to look at the latest and greatest hardware, when the end of sale is, and how long Cisco will support their product. Because, for example, in the event that you have a bunch of these deployed in your environment and, we can see, six months from now, there'll be end-of-support, then I should be in the planning stages of replacing these within my environment. Because what you don't want is for you to have a serious issue with one of these voice gateways, you Contact Cisco support or Cisco TAC and the answer that they give you is, "Sorry, can't help you, that hardware is no longer supported."

4:12So, here we have a 2911. I actually have this in my lab environment. But, before we start diving into the ports, we want to talk about the operating system. So the 1900 Series, along with the 2800, along with the 3900, all run Cisco IOS. When we're talking about the 4000 Series, this gateway runs IOS XE. So Cisco's IOS has been around for a very long time, since the 1980s, and it's still being used.

4:42And today, the IOS is a monolithic architecture. So what does that mean? The IOS runs a single image and all the processes share the same memory space. So, here, we have no separation of CPU or memory processes. Within Cisco IOS, if there is a single process that crashes, then that voice Gateway would become unresponsive. So it affects the entire gateway. IOS-XE a modular operating system, so processes are run on different modules.

5:19Therefore, unlike the IOS, if one process or module fails, it does not affect the entire router or voice Gateway like IOS would. In face, IOS-XE wrote a copy of IOS, so that is why many of the commands are identical. And finally with IOS-XE, the data plane is separated from the control plane. So the data plane is really responsible for moving packets from source to destination. So, we're actually forwarding the IP packet.

5:49With the control plane, we determine which path to use. So this is where you would find your routing protocols you also build and maintain the routing table, we're going back to this voice Gateway, the 2911. We see here that-- I have a card right there. This is a voice WAN Interface Card for viewing. And this would be used to connect to a PRI. So, maybe I have a T1 or an E1 circuit. So, within the US, you will have a T1, which is 24 channels, each channel being 64 kilobits per second.

6:25Now, if you're outside the US and Canada, you may see an E1 that gives you 32 channels. But, in this example, we have our voice Gateway here where, one side, we have the IP network and, on the other side, we have connectivity out to the PSTN and that voice Gateway bridges both networks together. Next we have some additional ports here. So, here we have an FXO card and on the other side we have a FXS card. So, with the FXO or Foreign Exchange Office, this gives us our connectivity via the analog lines to the PSTN.

7:01That's how you would connect or integrate these analog circuits into your IP network. You may have four lines going to this FXO card, each line having a phone number associated with that line. So if someone calls that number associated with one of those lines, PSTN would know to reach this voice Gateway as its destination, then somehow be routed to your internal IP network. You also have FXS or Foreign Exchange Station. So in one of our past jobs, we had facilities throughout the city and they had their own gas stations where the vehicles would stop and get fuel.

7:37These gas stations had an analog phone located near the gas pumps, so in the event they needed help. So, instead of having an expensive voice-over IP phone, we had these cards that connected these analog phones, so at that point, someone who needs help can pick up that phone and be routed to someone internally. Or, maybe you have some type of analog paging system or maybe a fax machine and those would be connected to this type of card to connect those devices to our IP network. So if you haven't worked with these cards before it's kind of difficult to remember what's what.

8:11So, I tell myself the FXO is used to connect to the central office. There's an "o" in office, there's an "O" in FXO. And then, for FXS, I remember that "s" stood for "station," so I told myself, it connected to a telephone, a fax machine or paging device, for example. So for every call that occurs in our environment, we have call legs and we have dial Peers. So, we saw that we have voice ports that we just went over and those ports connect us to telephony devices. Could be those analog phones that we spoke about, those gas stations, could be a fax machine, or it could be a connection to the PSTM.

8:49Or, we can't forget, you could be connecting to the ITSP. So, when a call is made, we have call legs all the way from the originating device going to our destination. So we can think of these call legs as static routes, if you're familiar with routing and switching. We have POTS that appears, that have to deal with a port of some kind. So, we define what port that we're going over, if that's a FXO, FXS or maybe a T1 or E1.

9:19Then, we have a voice over IP down here where the destination is an IP address. So, that's how we distinguish between the two, between a POTS dial peer and a voice over IP dial peer. So, when a call is made, CUCM has a call leg from itself to the CUBE, this example. So, in the perspective of the CUBE, here's the inbound dial peer, then that call has to go out the ITSP. And then, we have the outbound dial peer. And then, on the other side-- so, this could be a company A, company B-- it receives a call from the ITSP.

9:55Here's an inbound call to that CUBE. And then, we have an outbound call to that customer. So, without peers, we're specifying the origin and the destination of each call leg. Then, we can define certain characteristics with each dial peer. And we're going to find things like the CODEC they use. Also, can find the DTMF to be used for the call, how to interpret those digits that are being pressed. So, call leg is, really, a logical connection between two endpoints.

10:25There's a call leg between CUCM and the CUBE. And then, from CUBE out to the ITSP. As we solve the destination, we have the inbound call leg going from the ITSP to the CUBE. And then, the outbound call leg from the CUBE to the PSTN. So that, by itself, is four call legs alone. We have two here. And then, we have two at the destination CUBE. In this video, we talk about voice gateways, the different types of voice gateways that are available today. We also talked about some of the ports that are being used on these voice gateways to communicate with these analog lines or circuits or out to the PSTN.

10:58And we also had a brief discussion about call legs and dial peers. For that, I hope this has been informative for you. I'd like to thank you for viewing.

0:00[MUSIC PLAYING] So in this Nugget, we're talking about Cisco converge networks. We'll talk about the benefits of having a converged network. And also, we'll touch upon quality of service. Let's jump right in, and talk about Cisco converged networks. So converge is defined as moving towards the same point or coming together. Now, let's talk about an environment here that's not converged. So we can have a network here for our data.

0:32So these may have our PCs, our servers, printers, for example. But these are your traditional devices that you would find in a data network. You might have a separate network entirely for your voice. So you may have some analog endpoints in your environment, or maybe you have a PBX. And the infrastructure for your voice is completely different than for your data. So you have different cabling, you may have different components that manage and control that voice system. So these may be non IP based. You also may have the same with video.

1:03Maybe you have different video endpoints throughout your infrastructure. And you also may have different components along with different cabling, but again, the takeaway here that it's not IP-based. So you may have your staff or folks that manage the IP network. And then you may have a different staff that manage the phone system, or you may not. You may contract the maintenance of that phone system or voice system to a different person. So maybe Sharon here, is the one you call when there's a problem with the voice system, or you need to make a change of some sort, she comes in and she manages the phone system.

1:39You may have different folks that manage the video environment, or you may have someone else, like maybe Harper, for example. She comes in, she's a contractor also, just like Sharon, manages the video system for your environment. So we can see here, there's three different networks for different services for your business. So all these networks are maintained separately, and may have different technologies or protocols in order to provide that voice and video service. So the answer here, if you have a converged network, if you have data, voice, and video, those services can be provided on the same network, the same data network.

2:16So these services are all provided via the IP network. So now you have your desktops, your laptops, your servers. Now you have your endpoints. When it comes to phones and video, you have your tablets. You may have some time clocks in your environment, where people punch you in and punch out. But the point being is all these services are not provided on that same network. So now your infrastructure is no longer separated into different networks, it is combined.

2:47Now all these services can be managed by the same team, your IT staff, and no longer do you have to have folks with specialized skill sets, or those contractors, having to manage those other services for you. Since we have an IP-based network, cabling is a lot easier because we're all going toward wired network cables or possibly wireless. When choosing a vendor, now you don't have to choose one vendor for data, one for voice, and one for video, you could just choose one vendor like Cisco.

3:17And Cisco can provide all these services for you and your environment. It also becomes easier to deploy services. Because now when you're having a meeting-- I don't know why the guy is jumping and doing a split-- but now you can have a meeting with your staff. And now we can talk about deploying data and other services like voice and video to maybe a new site. And that communication becomes a little bit easier because all these services are IP-based. They all go over the same network. So having a converged network is extremely beneficial. But now we have additional traffic on our network.

3:49So there are certain things that we have to keep in mind. We have to keep in mind the amount of bandwidth that we have or were expecting to use. So here's our speed test for my home internet, and you can see I have a pretty good connection, but I'm getting 500 megabits up and down. So upload and download. Notice I said megabits per second, not megabytes. So with megabit, we are measuring the speed. So in this instance, the speed of my internet. With megabytes this is a measurement of data.

4:20So it could have a file that is 10 megabytes in size. So what's the relation between the two? So eight megabits is equal to one megabyte. Now these terms are often misused. So let's go over a few examples. For my connection, it is 500 megabits per second. So here we have converted this to how much data I can transfer per second. And let me bring over my calculator. So I take my 500 and I divide that by 8. Why? Because there's eight megabits in one megabyte.

4:53This gives me my value of 52.5. So if my internet was 500 megabits per second, that's how much data I can transfer per second. And also we see here how much data that would be per minute. So what happens if we have a slower connection? Let's say our site has 100 megabits per second. So that would equate to 12.5 megabytes per second, and over a minute I can transfer 750 megabytes. What if my connection was 25 megabits per second? I can then transfer it about three megabytes per second, or 188 megabytes per minute.

5:28So now we know with megabits, we're talking about how fast our network is, then we have megabytes, which is the amount of data. So we have an example here. Let's say you're downloading a Linux ISO, and the ISO happens to be 2 gigabytes in size. And 2 gigabytes is actually 2,048 megabytes. So going back to my connection of 500 megabits, that would take me about 32 seconds to download. If my connection was 100 megabits per second, it would take me about two minutes and 43 seconds. And if I had a 25 megabit connection, that would take me 10 minutes and 55 seconds.

6:01So all this to say, the a faster connection you have, the more data that we can transfer. Let's go back to my calculator. We'll take our 2 gig file, and we divide that by the transfer rate of 62 megabytes per second, and that's based in my connection is 500 megabits per second. So do that. 62.5, hit equal, and there you see my 32 seconds right there. There are some websites out there, in the event you don't want to do this calculation. So we take our 2 gig file. Let's say our connection was 500 megabits per second.

6:34And then we have our time to download that file. Next we have our delay, also known as latency. It's the time it takes for a packet to get from source all the way to destination. So for voice, how much time does it take, once I speak into the receiver, for that sound to get to its destination? And for this, the value for our latency or delay, should be 150 milliseconds or less.

7:06This is one way. Let's go to our command prompt. Let's do a ping to cisco.com. And here we see our round trip times. So when using ping, this is actually giving you the round trip time in milliseconds. And going out to the internet, that's pretty good. If I were to ping my CUCM server I have deployed in my environment, you see that at that time is a lot less. So here we have our times out on the internet, and our times internal to my network.

7:36So in the event we have noticeable delay in our environment, that delay in conversation could have people talking over each other, which of course, can be very frustrating. And we have jitter, which is also delay, but this is delay not from the source to destination but between packets. So let's say we have phone A, and we have phone B. So phone A is sending packets to phone B. Let's say the first packet takes about 10 milliseconds to reach the destination.

8:08The second packet takes about 20 milliseconds, then we have some congestion. So a third packet may take 50 milliseconds. So this jumps to about 75, maybe 100 milliseconds, and then jumps back down to 10 milliseconds. So here, the delay between packet one and two is 10 milliseconds. The delay between two and three, 30 milliseconds, three and four, 25. Then between four and five is another 25, and then here, we have 90. So here, jitter is the delay between packets, and our goal is to be 30 milliseconds or less.

8:43And here we see a little bump. Overall, with a lot of packets being sent, you definitely don't want to see this above 30 milliseconds continuously. If you have congestion within your network and see that, you may have distorted voice, or your users complain of distorted voice, along with gaps of silence. And then we have packet loss. So when talking about packet loss, we're talking about the number of packets that were received compared to the number of packets that were sent.

9:13So for example, if we send 100 packets and only 95% were received by your destination, then this will give us a 5% packet loss. And this often happens when there is congestion on the network. Now our goal here, for our voice and video environment, is to have a packet loss of 1% or less. So not knowing our goals for our voice and video environment, how do we ensure that we maintain those requirements? And the answer would be quality of service. Now, quality of service is used when there is congestion on the network.

9:47So let's say you have these packets attempting to leave our network, and there's no congestion. By default, the router treats these packets, or these packets, as first in and first out, or known as best effort. So I'm sure that many of us have been at airport. And before you board that flight, sometimes if you catch a super late flight, there's not a lot of people on that plane. So when it's time to board, they have this little door, or the boarding gate, and this door is what allows you to eventually board that plane.

10:17So when the agent tell everyone to get in line to board that plane, then that small group of people gets in line, and they all quickly are able to get on that plane. So this is first in, first out, best effort. Now, what if this was a popular flight and this plane, for example, could carry close to 200 people, and then someone screamed out, everybody get in the plane right now, and you had 200 people barge trying to get to that door? At that point you have congestion and pretty much mad chaos. The same thing that happens to our network when there is congestion.

10:49The router will have problems processing those packets, because maybe that interface on the router can only take so much bandwidth. So therefore, this presents an issue. So how does the router deal with the congestion? And that would be our quality of service. So let's talk about some of the data that's really important in our environment, first one being mission critical. These are apps that are critical to your business, and voice and video could be part of that list along with some other applications that may be crucial for your environment.

11:19Then you have transactional applications, maybe these are your databases that require real time responses. You have your best effort, maybe this is your web browser, maybe FTP. And then you have your scavenger, this could be your streaming, if that's allowed in your environment, or maybe your YouTube. But what applications fit under what category really depends on your environment. So maybe YouTube, for example, is a big part of your business. So that may be a mission critical app for your environment.

11:51So email may be mission critical for you, or it may be best effort. The reason being because a lot of your business is not conducted via email. So it's different for every environment. So as we stated by default, the router will treat that traffic is first and in, first out or best effort. And the packets come in, then those packets are then transmitted. But what if we have congestion in our network? Then the router will queue that data, so the data is waiting to be transmitted. But eventually the queue may get full, especially if there's some really bad congestion on the network.

12:24So what happens to that data that's not able to make it into that queue? That data is then dropped therefore impacting some services within your environment. And if we're dropping voice packets, then your users may complain about poor quality, which is a situation that you don't want to be in. So with quality of service we can do things like mark and classify traffic. So with voice, we can give this a tag, and I'll give it a letter. A tag of a, for video, we'll give it a tag of B, and then for data, we'll give this a tag of C.

12:54So we're marking or classifying this traffic. And these markings help identify this traffic as it crosses our network. So going back to our example at the airport, let's say that when you purchase your ticket, and this is the case with many airlines, one ticket says zone A, the other ticket says zone B, and one ticket may say zone C. So here, we are providing a marking on that airplane ticket that identifies what zone that they're in. On that, we can prioritize traffic, and why do we want to prioritize?

13:25Because if we affect our voice and our video, people will either hear the poor quality. So you might have chopped off words gaps and silence. At that point you can't really have a conversation with the other person. If we're dropping video traffic, they also might have choppy audio or their video may freeze, or-- but the end user would definitely see an impact there. And then with data. This could be maybe web browsing. So they might not see a hiccup there because with web traffic, they may see that page take a little bit longer to load, but really wouldn't see it as service impacting, as you would see voice and video.

14:01Because the reality is people are more patient for website to load compared to their audio or video being choppy, and not being able to have a conversation or a meeting. So I might say that my A traffic goes out first, has the highest priority. Our B traffic is next in line, followed by my C traffic. So we've marked our traffic. We now prioritize our traffic, and we do this via policy that we create. Then at that point, we will queue the traffic appropriately. So in the event we have congestion with our network, our voice traffic will be processed first, they have a higher priority, followed by our traffic, which is our video, then followed by our data traffic.

14:40We potentially may still have traffic drop, but at least we're getting priority so voice and video are less likely to drop than other traffic. Going back to our airplane example, we assign zone A, B, or C, to our customers tickets. And then the agent will load the zone A folks first on that plane, followed by zone B, and followed by zone C. So we're prioritizing the traffic there. So when it's time to board that plane, all the A folks get in line, followed by the B folks, then followed by the zone C folks.

15:11So this is our queue. So what happens if we have someone that comes in late, and they actually have a ticket with zone A? They could walk up here with many people online, and say, hey, I have an A ticket. So, therefore, I have priority. And I'll stand right there. So in this Nugget, we discussed converge networks, and that's when we're taking all of our services and combining them to a single network. And we also talked about quality of service, which we have to take into consideration, especially when we're introducing other services like voice and video into the IP network.

15:42And quality of service allows us to give that traffic special treatment, prioritizing that traffic, making it less likely to be dropped in the event that there is congestion on the network. So in the event you want to know more, go deeper within quality of service, there's excellent content here at CBT Nuggets that will definitely help you out. So that I hope this has been informative for you. I'd like to thank you for viewing.