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The next section were gonna take a look at is a brief history of Telephony
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The first thing we'll look at is a traditional (PBX) overview
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now a (PBX) or a post branch exchange
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was something that was originally created to emulate
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a (CO) or a Central Office
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so this was something that
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as telephone systems began to grow in size especially in corporations and enterprises
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the need for emulating the actual Central Office who used to just deliver a lot of analog lines to a company
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became more and more apparent people needed to connect to each other
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and didn't wanna necessarily have to pay the telephone company or go through
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the process of placing a full telephone company call
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just to contact someone in the office next to them or something similar to that
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not to mention that a lot of features
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were becoming more and more available
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by the central office and
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companies wanted to be to emulate these themselves
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and possibly potentially develop even more features and of course that's what's happened
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so, the private branch exchange
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came about and companies would put these in,
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in order to be able to as we said emulate a lot of these Central Office type features and components
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so the two main sides of a (PBX) if you will,
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if you'll consider the (PBX) sort of split in half
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is we have the station side sometimes called the line side and then we also have the trunk side
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and we also have additional accesory type cards
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for instance things such as voice mail or (IVR) either built in
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to the card that plug directly into the (PBX) or else uh was an extension where a connector came off and plugged into an external third party component
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for the station or (line) side these were line cards that were connected to analog or digital phones
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and the idea of a digital phone this was something that the phone itself would take and sample the audio
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from the speaker from the caller
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and it would convert it to binary ones and zeroes to a digital format and transmit that across the line
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versus an analog phone which would simply take the electrical impulses pass them on down the line and the (PBX) would do that switching and possibly transition to some sort of a digital format if they were going out
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maybe to other either another digital phone or some sort of a digital circuit like an ISDN PRI
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and typically a digital phone set was proprietary to the vendors, so you typically couldn't just buy any sort of digital phone set lets say from Nortel and use it on a Siemens
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or an Alcatel or a Lucent type (PBX)
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as far as the trunk side we would have trunk cards that were connected to the local Central Office
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so that is the local phone company the carrier, also sometimes known as the PSTN the Public Switch Telephone Network
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or possibly to another PBX maybe in the same location if someone had outgrown their standard PBX size
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although they would typically just upgrade to a larger size but that was one possiblity
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especially if there were some sort of a migration going on from one sort of vendor of PBX to another vendor, or else across long distance least lines that a carrier would provide so that they could trunk and connect multiple PBX's
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for a large company with multiples either multiple sites in one city
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or even spanning multiple cities
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so some of the disadvantages to the PBX architecture was that all voice traveled over a dedicated seperate wires to the PBX and I say seperate, seperate from the data network
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so many people when voice over IP first came about thought that this was one of the advantages to having the PBX architecture that everything was seperate and we'll certainly talk in a moment about some of the quality issues that can come from combining or unifying
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voice along with your data networks
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however, one of the big disadvantages was multiple wires needed to be run some specific to voice some specific to data or at least labeled and seperate
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so the cost of infrastructure went up, as well as maintenance. Also all voice inside of a PBX must go through a single
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Time Division Multiplexing or TDM as we'll call it from now on backplane so all of these line cards and trunk cards and accessory cards that we just talked about
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those all plugged in to a TDM back plane.
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Now the issue with this is that this can cause congestion at peak times because the back plane cannot support enough TDM channels
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so think of a virtual channel if phone 1 goes off hook and tries to call phone 2, that's gonna set up some sort of a virtual channel and there needs to be enough bandwidth if you will
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or enough back plane TDM channel or back plane or slots for each phone call
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to be made however there were almost never PBX's that had enough TDM slots or channels to support every single phone that they possibly could support from a theoretical design and you know
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building stand point so in other words how many phones you can theoretical add to the system there were no PBX's that could handle that many concurrent calls
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so that was a blocking architecture, what was known as a blocking architecture versus a non-blocking architecture
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that is to say that you could over subscribe the phone and CO trunk circuit provisioning in the design
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but you could not necessarily support that many calls going on at the same time one for every device that you have
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So taking a look at analog voice circuits, we've got a few different types of analog voice circuits
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some are known as FXS which stands for Foreign eXchange Station some are known as FXO which stands for Foreign eXchange Office and some are known as E&M which debatedly either stands for Ear & Mouth or else Earth & Magnet
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some people call it either one for an FXS or a Foreign eXchange Station important bit to remember is the "S" the Station
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this is the type of connection that you would use to connect to an analog station or an analog phone itself,
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this uses a two-wire signaling for tip and ring in an RJ-11 port
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it as we said typically connects to an analog phone or a fax machine although this can connect to an FXO port
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so FXS and FXO are like male and female in terms of electronic connections
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in terms of if we want to have two things connected to each other from an analog stand point
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on a phone system FXS and FXO can be connected together not that they necessarily should be
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but if you are on like let's say on a low budget and you want to do some sort of a PBX inter-connect maybe your connecting a newer
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unified communications system to an older PBX or an older what's also called a Key System Unit, which is a much smaller version of a PBX
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typically for very small companies under a hundred phones or maybe a hundred and fifty phones something like that
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you could use an FXS from one side from one switch from the PBX per se and then an FXO from the unified communications side from the voice gateway
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or vice-versa so this is one way of inter-connecting phone systems, it does have potential drawbacks that we could take a look at such as supervisory disconnect something that you'll probably learn a lot more about in C voice
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in the CCNP voice course and specifically the C voice course with in the CCNP voice certification but these are two things that can go together
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FXS typically also provides or not typically but does also out of necessity provide power so for instance negative 48 volts DC ringing voltage
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by the way so i've personally been working with Telephony for some time now
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and, going on well really if I consider all that I grew up doing with my father working at Bell laboratories AT&T Bell laboratories before it was renamed Lucent and then before their R&D Research & Development
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Division all that almost shut down but he was working there since I was about five years old and so he was also a patented adventure so I would have oscilloscopes and all sorts of fun things at home that I got to play with and learn it from
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learn when I was a young kid, but anyhow even professionally I was working with Telephony probably twenty years ago
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and one of the things that we used to do to new people (and I say we because well it happened to me when I first came to work at this company),
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as a, one of the companies that I went to work with was a 911 so in the U.S. emergency services provider, what was called a PSAP provider or a public safety answering point
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and we serviced a number of police and emergency type answering point systems, phone systems all over
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Probably about 500 systems in about a 9 state radius, but anyway when I first came to work for the company they did this to me and so in turn of course we would pass along the
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nice welcoming as a new employee to all the other new employees but if you ever want to see someone really jump pretty high
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tell them that, that testing to make sure that ringing voltage is very similar to testing a nine volt battery where you know granted probably not everyone does this with a 9 volt battery but in the U.S. we have these square (my wife calls them 'square' batteries)
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but these 9 volt batteries which both have the positive and the negative terminal on one side and if I don’t have a multi meter readily available I’ll just stick it on my tongue and I’ll see how much of kind of a chemical or electrical shock
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I’ll get from it's not very bad at all you know if it’s a pretty new battery then you’ll get a pretty decent oh wow you know little decent jolt and you’ll know that that’s a good one well we would tell that we would tell all the new recruits
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or I was told my first time that testing ringing voltage was very similar to testing a nine volt battery you would just stick your tongue on this and
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if you felt a little jolt just a tiny little one it didn’t hurt then you know you were getting ringing voltage and if you didn’t hear it on the analog phone it meant that the analog phone was broken and not necessarily the device like the PBX
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or the CSU oh sorry KSU that was providing the ringing voltage because those line cards would go out and stop providing ringing voltage quite often we’ll have to replace them
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well let me just tell you negative 48 volts dc is not like positive 9 volts dc,its one of the most abrupt shocks that you’ll ever get anyhow I jumped quite high and very kindly passed the initiation on
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but anyhow back to our learning so also an FXS port provides call progress tones such as ring back and ringing so when you go off hook and dial some ones phone or someones extension someones phone number
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you hear ring back the other side here’s ringing if you were on the fxs port on the analog phone and you had made the call that was you or the calling party then you would hear the ring back
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if you had not made the call someone else had made the call to your station then all of a sudden you will hear ringing or alerting also known as alerting and it also provides dial tone
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so this acts like an fxs port acts like the CO to the analog phone and in a digital system we would have digital FXS ports we have something similar in unified communications we still of course have FXS ports because we still have analog phones
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whether it be an elevator phone or a maybe a loading dock shipping area type phone or you know something of that nature or fax machine
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but in terms of IP phones and specifically when we look at unified CME or communication manager express we’ll see that we have something similar to what the pbx has had for their digital the older analog pbx’s had for their digital phone system
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and those were known as digital FXS port in unified CME we have eFXS ports or electronic FXS ports and those were what connect to our ip phones
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but they are still station FXS or station so fxo or foreign exchange office just remember that the O for office is what connects to the central office so an fxo is the type port that we would use to connect an analog loop start or ground start
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CO provided line to our pbx whether it be our old traditional pbx or our unified communication manager pbx and this again uses a two wire tip and ring still on our J-11 port
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as we mentioned this connects typically to the CO although it can connect to an FXS port it provides things like supervised disconnect and it supports caller ID
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so then we have an E&M port and as I mentioned either ear & mouth or Earth & Magnet and this uses an actually a 4 wire tip and ring and Earth & Mouth so the middle air are the tip and ring and this is where actually the analog voice goes through
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but then the outside are earth & mouth or ear & magnet these are outside pair are the signaling pair to an RJ-12 port so this is typically used for things like a ti-line to another pbx reason being we not only have analog voice but we also have signaling pairs
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we have a pair of wires that can handle a lot more information in regards to signaling than just a standard FXS or fxo where we might be passing forms of signaling on the same wires that are carrying the actual voice
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Another this thing that sorry another thing this can be used for is a music on hold source so maybe we have a live feed from a radio or a Musack is a type of company provides a live feed or maybe we have a paging source
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and so typically those will use the inside pair sometimes some music on hold or paging systems are designed specifically for E&M and so that they'll use not only the inside pair but also the outside pair for signaling
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but a lot of times music on hold or paging can be connected to the inside pair of an E&M that would be the proper way to set that up
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So taking a look at digital voice circuits in comparison to analog digital circuits carry information in what are known as channels and there’s two main types of channels we typically have bearer channels or data channels
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and before we talk more about the channels we’ll look at well let’s just go ahead and mention real quick the bearer channels are what bare the actual voice so the bearer channels are what carry the audio along
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data channels are used exclusively for control data or setting up tearing down passing information such as calling party number or called party number
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things of that nature and a lot of other information but one other thing to note is that bearer channels so data channels only carry information about the call data
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but the bearer channels can carry only voice or on certain types of circuits that carry voice and control data about the call so let’s take a look at the two main types of digital circuits
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one is known as CAS which stands for channel associated signaling and one is known as CCS or common channel signaling
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we’ll take a look a little bit more of a look at those on the next slide and as we’ve already mentioned time division multiplexing or TDM is used on all circuits
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so many voice conversations are each sampled and then as each voice conversation has many samples for each conversation
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it’s important to keep those conversations separate but we also can’t just have one conversation
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monopolize the entire line, so instead what we do is we cut up each of the voice conversations into chunks or slices
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and we interweave them with one another because our digital circuitry can handle a lot more information than just a single call on a single line
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this is very much like the ability for a single pair or actually a couple pair of wires copper wires to be able to carry ten megabit of information, ethernet and data. And then later electronics on either end to get smarter
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and with the same exact copper carry information at ten times the speed at a hundred megabit such as fast ethernet and then maybe a little bit of improvement to the quality of the copper or the lines but still copper
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still two pair and we can carry information ten times that at gigabit speeds, so the same idea is there with voice although these voice circuits haven't evolved as nearly as fast as data circuits have
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and we have the ability to have two pair of wires carry not just one conversation or two conversations but 25 or 30 conversations and so forth and that's what we'll look at with these TDM circuits
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But the only way that we can do that is if we have them carry all these circuits in parallel so all at the same time beside each other and really the only for that to work effectively is if we interweave them back and forth
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but we have to have control data that controls which conversation is dealing with which and one of the primary ways that we do this is with something called muxing or multiplexing and we do it by dividing up based on time
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so it's very important with time division multiplexing that both sides of the circuit agree on the same clock so clocking is very important to digital circuits if you don't have clocking set-up properly in a master slave type relationship
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where one side, with no questions just says okay whatever you say the other side says is the time, the definitive time that's what we will agree on and use without some sort of common or agreed on synched up time
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then we begin to have what are called slips in our digital circuits where voice information can begin quickly to degrade or fall off or will loose track the digital circuits will loose track of whose conversation was
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and voice packets or not really packets but voice conversation voice time slices will be dropped.
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So let's take a look at Channel Associated Signaling or CAS, so voice and data in a CAS circuit are both carried on bearer channels so there is no seperate data channel with Channel Associated Signaling
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the idea is that the channel that's carrying the voice that's associated with the voice is also carrying the data, it's associated with the voice channel or the bearer channel so the data is associated with the bearer channel or the voice channel
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so each bearer channel is robbed of certain bits that would otherwise be used to carry voice in order to transmit the data for that call alongside the voice this is one of the reasons why or Channel Assoicated Signaling sometimes call Robbed Bit Signaling
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because there are certain bits that are robbed from each channel in order to carry that data an example of this would be a T1 Wink Start or possibly an E1-R2 those are both examples and forms of Channel Associated Signaling or CAS
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Then there's common channel signaling so this voice alone is carried on the bearer channels and then data about the voice is carried on a single channel or a common channel so that is to say that the data that is controling all of the calls
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is carried on a single channel that is common to all of the other channels so regardless of what type of circuit you look at your always going to find data for a T1 common channel signaling which is ISDN PRI
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your always gonna find that carried on I guess I could say PRI or BRI with the PRI we have 23 bearer channels and one data channel but it's always gonna be carried on channel and depends on whether were counting chononacally or not
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which is to say if were beginning our count at number 0 or if we are beginning our count at number 1 but assuming were beginning our count at number 1 voice or data for a PRI for a T1 would be carried on channel 24 always
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Data for a E1 would always be carried in the middle of the circuit and it's gonna be carried on channel 16 so there's 32 channels on any one and data is going to be carried on channel 16
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or a BRI or a Basic Rate Interface we only have two bearer channels and one data channel so we gave the example of this being the T1 or E1 ISDN PRI or BRI
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So moving on let's talk about a circuit switch network, the circuit switch network involves atleast two nodes so for instance phone to phone or phone to CO etc. And they establish a dedicated channel or circuit in order to talk
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and the circuit remains connected for the entire duration of the call, the circuit really acts as if the nodes were physically connected to each other with an electrical circuit so it again it acts it emulates this
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they are connected with some sort of circuit but it's not necessarily that they're directly connected to each other, they maybe switched to be connected to each other but they may also be time division multiplexed
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and so therefore there isn't just one wire that is responsible for connecting two phones but one wire might be responsible for connecting many phones but it acts as if there's a direct connection between this
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the Public Switch Telephone Network or PSTN as we'll often refer to it, is the largest circuit switch network in the world. If your not very familiar with voice and your just getting started
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you could think of this sort of like the internet but instead with many telephone company or Telco switches rather than routers and switches or IP packets and they're all connected together with TDM circuits
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using something called the SS7 protocol or Signaling System 7 and this is how most modern PSTN couriers connect to each other and connect their own switches to each other. Now I might even back up and say most modern carriers
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are more and more if not close to all interconnecting their own systems and even with other carrier systems through IP rather than just through SS7 and some are even back calling the SS7 signaling over IP
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but just to look at traditional approach if you'll allow me, most carriers and most PSTN's are connected or the PSTN is connected together with SS7 signaling.
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So advantages and disadvantages of circuit switch networks some of the advantages are well we have a constant connection, so you guarantee that if a channel is available every bit from the voice call will arrive at the other side nothing will be dropped
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there is no packet loss or no dropping, assuming that we have good clocking as I mentioned if we don't have good clocking then there could be drops. But assuming that clocking is good then we guarantee that every bit reaches the other side
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the bit rate of the call and the delay, so however long it takes that bit to get from one side to the other those both stay constant. And so therefore quality is inherently excellent
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Some of the disadvantages of a circuit switch network there's not always enough channels available, so we have a blocking architecture. Sometimes especially if you happen to depends on how old you are
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and if you remember you know even maybe 20 years ago maybe even as early as 10 years ago, but certainly 20 years ago before the advent of massive growth in infrastructure of both PSTN, TELCO's and cellphone providers or
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mobile providers and even IP or carrying voice 20 years ago if there was a major catastrophe somewhere and you went maybe say a tornado ripped through an area and people tried to get on their home phones their analog home phones
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or even a public phone and make a call you could expect at some points to hear something similar to you know all circuits are busy now, all circuits are busy please hang up and try your call again later this is a recording
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and so you'd get sort of an enunciator type recording telling you that all circuits were busy and this is because we had a TDM back plane architecture even for the PSTN couldn't simply just couldn't handle every single person
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even in a city much less a state or a country or the world making a call at once so that's one of the disadvantages, another disadvantage is that while the bit rate stays constant it's also limited so there's no or atleast little chance for newer technology
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to improve sound quality wasn't really that there's no chance for newer technology to improve sound quality but people thought that for the most part until IP came along the voice sounded just fine over a standard PSTN type connections circuits and trunks
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and so really there wasn't a lot of development in the way of technology for improvement in sound quality and so the bit rate has for the most part on devices on analog devices and even digital trunks that carry
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or convert analog voice it stayed the same, part of this is due to the fact that analog circuitry FXS ports and things like that an FXSO ports would literally have to be upgraded in order to provide you with a
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the actual port, the actual technology and the analog phones would have to be upgraded in order provide you with a better quality. So another disadvantage was that you would have to bond many channels together to allow things like video
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so we had you know even as early as probably 10 years ago and certainly 15 and 20 years ago we had video conferencing but instead of being over IP atleast 20 years ago and 15 years ago it was over ISDN
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and you would typically connect atleast 2 ISDN channels or maybe a BRI 2 ISDN channels to get and each ISDN channel can carry a 64 kilobits worth of voice or video or data each B channel and so if we
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connected a BRI which is 2 bearer channels plus the data channel we could have a 128 kilobit video call or data for back up data. So if we you know it was very common for video conference systems to have 3 BRI's running into a site
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or something like an ATLUS 550 or an ATLUS 800 that would be there to split certain channels out of a PRI you know maybe a 6 B channels split off into 1 video conferencing room and another 6 into another conferencing room for another video conference
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and then the remaining 11, 12 would be one of them would be the data but the remaining 11 voice circuits into let's say a PBX or a smaller KSU or something like that but we'd have to bond these channels together to allow for video
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So looking in contrast to a circuit switch network looking at a packet switch network, this is a digital network that transmits data irrespective of what the data contains as a payload the content
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0:33:07
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into blocks of data, called packets so layer 3 devices we'll just go ahead and take this out of the way layer 3 devices carry packets and then they encapsulates those into layer 2 frames
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0:33:19
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sometimes you'll hear people, you know you might even hear me on some videos accidentaly erroneously speaking of packets being carried through a switch layer 2 data switch.
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0:33:32
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Now the truth is the packets are being carried through the switch, but they've been encapsulated into layer 2 frames so when were speaking layer 2 regardless of whether it's an ethernet switch
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0:33:44
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whether it's frame relay or ATM were typically talking about frames I guess actually ATM carries cells but if were dealing with a layer 3 device they're gonna be carrying packets
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0:33:58
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and then each device makes decisions independent from previous layer 3 routing or layer 2 switching device that determines where that packet or frame respectively should be sent
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0:34:11
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and the internet is obviously the largest of these packet switch networks but of course there's many private enterprise networks as well.
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0:34:19
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So looking at advantages and disadvantages of Packet Switch Networks some of the advantages are that bit rate stays constant, it's not limited however so newer technology often improves sound quality
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0:34:42
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So that is to say that we don't necessarily have to upgrade an entire type of a port especially with an electronic type of a port in order to improve sound quality, we typically do have to upgrade the phone to be able to support that newer codec
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0:35:01
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or the sound quality but a lot of times if the phones have been built with enough infrastructure enough memory or processing power or what have you then they can be done over the IP protocol with a simple firmware upgrade
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0:35:18
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It's a lot easier to increase bandwith for more features such as video and something called quality of service that you'll learn quite a bit about as you go on, on your voice studies allows us to overcome all of these disadvantages
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0:35:35
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so some of the disadvantages that quality service allows us to overcome, well first of all no dedicated connection as with circuit switch networks so that is to say that the delay is not guaranteed to stay the same
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0:35:51
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and probably and normally doesn't and that there's no actuall inherent guarantee that the packet will ever arrive so a layer 3 packet going over some sort of layer 2 medium could potentially be dropped
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0:36:06
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now quality of services what gives us the guarantee that we can ensure that no packets specif to voice or possibly video that's no unified communications packets would be dropped
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0:36:20
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but without that there is the possibility for packet loss of course there's not always enough bandwidth available although it's much easier to add bandwidth than it is to add voice circuits and delay can occur at various router hops along the way
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0:36:40
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Also delay can and very often does greatly vary from hop to hop so it's not only that there is an unpredictable or sometimes unpredictable amount of delay, but that the delay can actually change from router or switch hop to hop
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0:37:04
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and this difference in delay is known as jitter, course delay could also be referred to as latency so when we have a variance in latency or variance in delay that is to say maybe we have a 3 mili-second delay at 1 hop
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0:37:20
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I really hope we don't have a 3 mili-second delay what I meant to say was a 30 mili-second delay at 1 hop and a you know 10 mili-second delay and another hop half a mili-second delay and another hop and maybe even as much as an 80 or a hundred mili-second delay
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0:37:44
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at a different hop all of these differences in delay are known as jitter. So looking at unified communications networks in a unified communication network everything flows over the same network as standerdized data
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0:38:01
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however quality of service keeps these UC packets prioritized so voice calls maybe a hardware based phone and or MAC or PC software based phone and then also video calls again based on hardware phones based on video conferencing IP video conferencing units
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0:38:20
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Telepresence things like that also software MAC and PC software based soft clients and dealing with conferences voicemail presence and instant messaging directory services and possibly even phone based patients
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0:38:42
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Looking at public providers unified communications systems can connect to traditional PSTN or even newer VoIP type providers, so a newer VoIP type provider such as an Internet Telephony Service Provider or ITSP as we'll typically call it
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0:39:00
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and they typically use either the SIP protocol which is the most standard these days or possibly the H.323 protocol for older providers for both ITSP's and we'll take a look at these protocols in a little bit but they certainly have the ability to connect to traditional PSTN
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0:39:21
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through analog through digital so they stil have FXO they still have FXS they still have E&M and they have the ability to do channel associated and common channel signaling digital circuits.
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0:39:34
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They can connect and interconncet with other PBX's again using all of these methods using any of the analog or digital circuits as well as IP, so just looking at voice over IP in general the idea is taking analog voice that is the kind that you're hearing right now
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0:39:56
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or video and sampling and encoding it into a digital value for transmission in an IP data payload, so we sample the audio we have specific type circuits that are, were gonna take a look at called DSP's or Digital Signal Processor's
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0:40:15
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that sample this audio and then they encode it into a digital value a 1 or a 0 for the actual hacketization into an IP data payload and then for transmission along the layer 2 or layer 1 medium
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0:40:32
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We use both a signaling protocol and a medium protocol to comprise the overall conversation and their typically transmitted seperately so that is to say the signaling protocols used to do things like set-up tear down and control information about the call
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0:40:54
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and when the signaling has been successfully negotiated that will result in media transmission media protocol actually does the sampling of the voice or video into something called a CODEC which we'll take a look at next
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0:41:10
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and is completely independent from the signaling and typically sent using UDP.
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0:41:20
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So a CODEC or COmpressor DECompressor CO from COmpressor and DEC from DECompressor is what's used to make the not really the acronym but the word formation of CODEC, and CODEC carries the actual voice to our video
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0:41:40
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actually does the sampling of it and the decompressing of it so the putting it back into an analog format for the listener a compressor is used in the transmission or sorry transmit direction so we compress the call and then we transmit it
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0:41:59
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and a decompressor is used in the received direction that is to say packet is received and then it is decompressed, CODEC is actually carried in the Media stream. Something that we'll take a look at called the RTP stream RTP stands for Real Time Protocol
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0:42:19
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And CODEC's include what are not limited to these are just the more common ones G.711, G.722, G.729 the iLBC the iSAC and we'll take a look at what these CODEC's have in common or differently most of them we'll take a look at what they differnt from each other
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0:42:45
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what they have in common is that they all carry voice and/or potentially video, so looking at the G.711 CODEC this is an ITU standard the ITU is the International Telecommuncations Union so this is an organization that's an
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0:43:02
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International standards organization that's been around for actually a research and development organization the ITU-T we had a dash T at the end that's the International standard's organization for the ITU
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0:43:16
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but they've been around for atleast a hundred and fifty years well maybe not a hundred and fifty years maybe a hundred and twenty something like that and dealing with voice and audio you know ever since the Edison and Alexander Graham Bell days
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0:43:37
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G.711 uses Pulse Code Modulation or PCM 64 kbs per second is the default rate it was really used to emulate an ISDN channel or Common Channel Signaling or CCS type digital circuit because ISDN digital circuits also use PCM or
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0:44:03
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Pulse Code Modulation at a 64kbs rate what was used to sort of emulate that it's got excellent audio quality and it's typically used for LAN connections certainly doesn't mean that's the only thing you can use it for
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0:44:22
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but that is where it is most of the time used and recommended to be used, the G.722 protocol is a lot newer than G.711 although it's certainly got it's years by now it again is an ITU standard it's optimized for wide band speech
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0:44:45
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so the difference in wide band being that there's a larger slot or a larger bandwidth a larger spectrum of the frequency that is allowed to be sampled and carried in the actual CODEC it again uses a 64kbs default rate
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0:45:05
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so it has a incredibly improved audio quality at the exact same transmission rate so no penalization at all for using it the exact same transmission rate as G.711 so it's got superior audio quality far superior in fact some people
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0:45:27
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i've installed systems for some people when they wanted to use they've heard about and wanted to use the G.722 protocol and it can be eerily good so some people say that it's actually too good and it sounds a little bit eery it sounds like the person is in the room with you
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0:45:49
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just too close for comfort for some people so they want to take it off but it's an amazing quality again typically to use for LAN connections looking at the G.729 CODEC this has an 8kbs default rate kbs per second.
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0:46:09
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It's a high complexity CODEC in the sense that the algorithm that is run is both CPU resource intensive and DSP so DSP or Digital Signal Processor is what's used in all audio products not just Voice over IP
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0:46:30
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but you know if you ever looked at a sound board or looked at a stereo, amplifier or receiver or a home theatre system all of these different types of products and many many many more use DSP's
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0:46:44
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Digital Signaling Processors anytime there's audio or video involved going from analog to digital typically it's a DSP specific ASIC Application Specific Integrated Circuit that you used to sample and compress or decompress and turn back into analog audio waveform.
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0:47:07
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But their very high complexity high resource it's used for LAN connections and the G.729 CODEC has variance so it has many different NXS's as we call them so things that were added on to the CODEC
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0:47:29
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all the base encoding is still the same but the ways about things are different the only two that we'll take a look at for right now are G.729A and G.729B so one of the things that G.729A gave us was instead of a high complexity CODEC in terms of CPU and DSP channel resources
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0:47:50
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it utilized the Medium complexity also with the G.729B variant were back to a high complexity in terms of CPU and DSP but what we have is the addition of voice activity detection or VAD and as you'll come to know VAD is bad
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0:48:12
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bad really doesn't stand for anything I just put it all in caps but VAD or Voice Activity Detection the idea was that in older and were talking 10 years ago even before, older days when we had a lot less band width available maybe we would have a
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0:48:36
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hundred and twenty-eight K CIR PBC for frame relay between two company sites maybe a headquarters and a branch site and maybe even some places still have limited band width and we wanted to use maybe some had 64K circuits
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0:48:56
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and we wanted to have the ability to have voice carried over these IP circuits, well with the addition of VAD or Voice Activity Detection the idea is basically voice activated so as the microphone begins picking up audio and the DSP's begin sensing audio
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0:49:17
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they go ahead and begin listening, sampling, compressing and transmitting that audio. But then when there's a silence in the audio so you know for that 2 seconds or 1 and a half seconds or whatever it was
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0:49:35
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that I didn't say anything that's where the DSP's that will sense that there's no audio, and they'll stop listening they'll stop sampling stop compressing and they'll also stop transmitting. So what this means is that there's on average a voice or actually IP network savings
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0:49:55
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so, it sounds good in theory and practice the problem was that it's not that they couldn't have gotten better at it, the problem that i'm going to enumerate here the solution was simply that we got a lot faster bandwidth and the need for this kind of
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0:50:13
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Voice Activity Detection and only transmitting when there was audio pretty much went away, but the problem was that if I would take a lot of pauses in my speech sometimes I do. Then what you would notice is that you would begin having clipping at the beginning and end
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0:50:34
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of conversations or just even of words or sentences, so if I took a lot of pauses in between it would begin sounding something like this, hear me talk and all of a sudden it would go, I can't really emulate clipping that well but it would clip the beginning
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0:50:54
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of a word or sentence and the end of a word or sentence off typically the beginnings a lot more than the end as that VAD or Voice Activity Detection became activated so that's why people got used to the phrase VAD is bad.
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0:51:08
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Now another part of the problem was that if you stopped sampling audio and therefore stop transmitting audio if all of a sudden you just hear dead silence in between communication on a phone the psychological effect is for the listening party to think that you've hung up
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0:51:28
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or that the circuit got disconnected and certainly with less reliable circuits in the earlier days circuits becoming disconnected was a common thing and so if there was literally no audio at all, you would think that the circuit would have been disconnected
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0:51:44
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and so what they had to end up doing was creating comfort noise, noise that would comfort the human psyche into thinking that the connection was still very much valid and there because it was and this comfort noise would be generated on either end wouldn't be transmitted
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0:52:00
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if it would be transmitted the whole idea of not transmitting when there was no audio went away.
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0:52:08
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So VAD and Comfort Noise Generation or CNG was what G.729B NXB or variant primarily provided. Newer CODEC's such as the iLBC or internet Low Bitrate Codec use 13.3kbps by default as again this is just the default rate
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0:52:32
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so this does not get into necessarily layer 3 or actually layer 2 header information this is just the default rate of the Codec itself this is optimized for narrowband speech and for lossy WAN connections.
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0:52:48
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So the idea was that narrowband speech so were just looking at a very small swath or spectrum of the available frequency so an available audio frequency the human ear can typically hear somewhere, this is debated on
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0:53:07
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and also not only debated on how much of either far end of the frequency spectrum for audio that humans can hear but then also as you begin to get older of course your ability to hear a lot of these frequncies begin to deteriorate on either end of the spectrum
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0:53:27
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but humans can typically hear between 20 hertz at the low end their 20 hertz not kilohertz but just hertz and really something as low as 20 hertz is felt not really heard but can hear from or feel, sense from about 20 hertz all the way upto about twenty-thousand kilohertz
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0:53:50
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oh I'm sorry 20 kilohertz, so twenty-thousand hertz so anywhere from 20 hertz on upto twenty-thousand hertz but with narrowband speech what were doing is were limiting it,
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0:54:02
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i don't remember the exact frequencies but it's somewhere right around the 1K one-thousand hertz or 1K one kilohertz spectrum so right around I think it's about 880 hertz on up to about 3 thousand hertz or 3 kilohertz
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0:54:21
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that range right there is typicall male female speech now let's not to say that male female speech doesn't have presence in some of the lower and higher frequencies it certainly does and that's why wideband CODEC like G.722 sounds so superior
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0:54:38
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to narrowband CODEC's but the idea is that most of the information that's needed to discern what a human is saying male or female can be got with the, can be reached or understood can be attained within that 800 to 3,000 per spectrum.
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0:54:58
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So in order to preserve on bandwidth sometimes narrowband sampling is taken as one of the ways to deal with that, and it also deals with lossy WAN connections such as the internet so the idea as if were transmitting over the internet
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0:55:17
|
there is no guaranteed quality of service atleast not until net neutrality wins out it's you know whether pro or against net neutrality we're not gonna get into that issue but at some point if quality of service makes it's way to the internet
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0:55:34
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again not necessarily good not all necessarily bad but you know there's pros and cons to both but aside from that argument, if quality of service makes it's way to the internet then we could potentially deal with a packet loss and prioritization of voice
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0:55:53
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and video, and i'm sure some happy medium will be reached sometime in the future but for the moment
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0:56:01
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and has it been for a long time the internet doesn't guarantee any quality of service so that is to say it doesn't guarantee that your voice will reach the other side there could be packet loss
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0:56:11
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and so the iLBC CODEC was one CODEC that was designed specifically to deal with the fact that we may not only have large variances in delay or latency so a lot of jitter but we also might have packet loss that is certain packets carrying
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0:56:30
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voice information may get lost altogether, dropped due to congestion and then we also might have packets that arrive in the wrong order and if voice packets arrive in the wrong order
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0:56:44
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there's typically depends again on the CODEC your using but typically not a lot of room for reassembly or reordering of those packets in the proper order so that the payload which is your voice ends up in the proper order
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0:56:58
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you know you wouldn't just hear words backwards but you would hear sounds and semblance and things like that, little inflections and tonalities and things in the wrong order so packets voice packets and video packets arriving in the wrong order can be disastrous
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0:57:17
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and the time that it takes to reassemble those in the proper order can be a big issue so one of the deal one of the ways to deal with this is with a CODEC that specifically buffers especially on the receiving end
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0:57:30
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so both ends have a receive path both ends have a transmit path but buffers on the receive and tries to deal with packet loss and especially packet reorientation or reordering as best as possible.
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0:57:44
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Another CODEC specifically and this is a much newer one than the iLBC even though the iLBC is a fairly new CODEC,
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0:57:53
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this one is specifically the iSAC or the Internet Speech Audio Codec and the idea with this is that it's an adaptable variable rate so it can be as low as 10kbps or it can be as high as 32kbps, now that's still half of what G.711 and G.722 are at 64kbps
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0:58:15
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But it could go as high as 32, this was again dealing with packet jitter or variance in latency or delay overband connections like the internet specifically for the internet as well as packet loss but it incorporates wideband speech
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0:58:36
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so it's sampling a much wider spectrum of the audio frequency.
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0:58:44
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So looking at the DSPs or Digital Signal Processors as we've sort of already eluded to and mentioned these convert analog to digital and then on the other side so that's on the transmit side and then on the receive side digital back into an analog audio wave form
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0:59:02
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as well as video as well. And so a DSP samples or listens to the analog voice or video it then goes through a process called Quantization which approximates a continuos range of sampled values that uses this after sampling
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|
0:59:22
|
it approximates these sampled values goes to the Quantization and then goes through a process called Encoding, converting the Quantization value into a digital value and this is where the CODEC comes in and in packetizes them into an IP data payload
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0:59:44
|
and then if finally does the compression, so it optionally compresses packets using hashing algorithms now this is compression is not always obtained in terms of IP or
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|
0:59:58
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TCP or UDP compression but we will take a look at some possibilities of IP or UP sorry UDP or TCP compression.
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1:00:10
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So looking at PVDMs or Packet Data Voice Modules these anytime we refer to a PVDM or PVDM2 or PVDM3. PVDM2 is just a second generation PVDM3 is a third generation etcetera and so on but these are Cisco-proprietary DSP's or really Cisco's implementation
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1:00:35
|
of industry standard DSP's okay so whether their texas instrument chips or whatever the actual chip set is these are Cisco's implementation not only of these DSP's but also putting them on a circuit board and adding a few other you know Cisco secret suit type proprietary things
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1:00:55
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But these are used in IP phones for voice termination so by the way whenever we say in a voice or Telephony world whenever we use the word terminate sometimes we are and you just have to listen in context sometimes we are reffering to
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1:01:13
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the call being torn down but most of the time that's not what were referring to, most of the time if were talking about a call being torn down just say hang up or tare down
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1:01:23
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but when we're talking about voice being terminated on either end what were typically talking about is it being stopped and terminated at some sort of point of demarcation
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1:01:35
|
and therefore utilize, so if a voice conversation is terminated between two phones then it is actually up and running and alive. But in our IP phones all the different types of IP phones that Cisco provides as well as just about every other vendor they're going to have DSP's
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1:01:56
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okay,so or possibly PDVM's as we might wanna call the for Cisco.
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1:02:02
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These are also used in VoIP gateways for voice termination so what we mean there is between an analog circuit and an IP leg of a circuit, were going to have the call terminated there on a PVDM so going from analog to digital and then back from or I should really say from analog to IP
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1:02:26
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and then back from IP to analog as well as on digital circuits so maybe a ISDN PRI just what type of signaling, that's right Common Channel Signaling so maybe an ISDN PRI coming in from a TELCO coming in from the PSDN
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1:02:48
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And then the call terminates into PVDM's and then is packetized into IP
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1:02:55
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this is also used in VoIP DSPFarms, so a DSPFarm is pretty much just what it sounds like a pooling together of these DSP resources or PVDM resources and this is useful for things like conferencing, Media Termination Point
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1:03:13
|
MTP is another way to say Media Termination Point, a place where we terminate media and then reoriginate it for many different reasons that we'll look at and then also possibly for Transcoding when we talk about Transcoding,
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1:03:29
|
Transcoding terminates media so it is a an MTP of sorts it actually can do everything an MTP can do but has the added ability to code or to have two different CODECs on either side of the call, so that is to go between these two different CODECs or Transcode
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1:03:55
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Looking at the RTP or Real Time Protocol, RTP is a Layer 4 protocol it rides atop of UDP also a Layer 4 protocol and it encapsulates all delay sensitive traffic, so delay sensitive traffic is are unified communications traffic are voice in our video
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1:04:18
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RTP uses the UDP port range from 16384-32767 and in some devices not very many anymore but some especially older devices of Cisco's would see this as 16384 space 16383 and that's because after the value of 16384 there are another 16383 ports that can be used
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1:04:55
|
so it is actually in itself a 16,384 port range protocol using UDP ports, okay but that range is the actual starting UDP port of 16,384 and the ending port number 32,767
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1:05:17
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Also with RTP for the Real Time Protocol is the RTCP or the Real Time Control Protocol, so this is a control and quality & statistics protocol that is sent along aside or "paired" with each RTP stream now this is not to say for every RTP packet there is an RTCP packet
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1:05:41
|
that would be too much to chatty but for every X number of RTP packets there will be an RTCP packet so a control protocol just dealing not with the actual set-up and tare down of the call that's the VoIP signaling protocols that we'll talk about next
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1:06:04
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Such as a H323 and SIP that we talked about but this is just dealing with the actual control and quality and statistical information of the actual media. Okay, it uses the same randomly chosen port as the RTP stream chose except for +1 so if an RTP stream
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1:06:26
|
let's say the transmit port that were transmitting to this port you know the other side chose the port of let's say 16,567 and they told us that so were sending all of our RTP traffic to the destination port of 16,567 is that what I just said?
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1:06:48
|
The RTCP will be 16,568 so we'll just use a +1 and I gave another example there written out
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1:07:02
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Also dealing with security by means of encrypting the voice data payload using the AES cipher or Advanced Encrypting Standard cipher then again this does not encrypt the entire packet as IPSec would just the voice or video data payload
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1:07:19
|
is performed by secure RTP or SRTP sometimes you'll see this actually written as a lowercase "S" and then upper case "RTP"
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1:07:30
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And this is done through TLS or Transport Layer Security, Transport Layer Security you might be more familiar with SSL Secure Socket Layer, SSL had version 1, 2 and 3 and then SSL went to version 3.1 they stopped calling it SSL and they began calling it TLS
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1:07:50
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So, Transport Layer Security is what carries along secure RTP or encrypted audio packets. So if we take a look at the encapsulation the way that this would actually work, from the right we see that we have our voice payload
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1:08:09
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and that gets encrypted with an RTP header the RTP header then gets encrypted this is Layer 4, gets encrypted with another side layer for UDP header then the Layer 3 IP header
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1:08:26
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and then we have whatever our Layer 2 protocol is if we have that header to be placed into a frame and sent.
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1:08:37
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So taking a look at VoIP signaling protocols, signaling protocol these are used to set-up, teardown and control information about the call such as 'Supplementary Services.' Supplementary Services such as call forward call transfer
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1:08:55
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old, busy, call redirect, call park, call pickup information about presence or status message waiting indicators (MWI's) and etcetera etcetera etcetera so these are known as Supplementary Services
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1:09:11
|
and all of this, all the different Supplementary Services plus all the you know basic call control set-up teardown this is all handled by the signaling protocol
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1:09:22
|
the most common protocols certainly for what we use in our Cisco unified communications network are H.323, SIP, MGCP and SCCP. So let's take a look first at H.323 this is an (ITU) standard, this evolved from ISDN
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1:09:49
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and ISDN use something called Q.931 if your new to Telephony there are a lot of acronyms and if you have done routing switching or if you have atleast have a decent data background hopefully you've been through the CCNA just the
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1:10:08
|
actual base CCNA the network and routing switching CCNA some would call it as that is a pre-requisite for CCNA and CCNP voice, your no doubt fairly familiar with acronyms but man do we have a lot in voice, so anyhow
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1:10:29
|
so ISDN and I actually wrote this backwards ISND but it would be ISDN, and ISDN stands for Integrated Services Digital Network but that use something called Q.931 and ISDN actually did have a Layer 1, Layer 2, Layer 3 component to it
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1:10:49
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didn't have higher like the rest of the IP OSI layer goes up to 7 but it did have 3 different Layers and Q.931 was the Layer 3 protocol Q.921 was the Layer 2 protocol so that were fairly easy to keep seperate Q.931 for Layer 3 Q.921 for Layer 2
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1:11:10
|
but this evolved H.323 evolved from the Q.931 signaling and specificaly from something called H.320 and H.320 was not an IP based protocol but this was the protocol where video conferences would utilize
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1:11:32
|
multiple B channels or bearer channels as I mentioned before bonding up to as many as 6 bearer channels in order to create a 384kbps video call
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1:11:44
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and so this is where H.323 got it's roots from, it is a Peer-to-Peer Protocol so Endpoints typically called Gateways are intilligent and they have independent dial plans
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1:11:58
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so this means that no registration is required and even to date if I have an H.323 some people won't say the dot if I have an H.323 gateway in a Cisco Unified System, lets say thats a router running the H.323 protocol
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1:12:16
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and by the way when we use the term Gateway, what we're typically referring to is IP some sort of IP or VoIP Endpoint that provides services to a Gateway of other services or circuits or numbers
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1:12:33
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When we use, Endpoint that could refer to something like a Gateway if we refer to a terminal or a node we typically referring to one device that has you know 1 or maybe at the max 2 phone numbers associated to it
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1:12:50
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but a Gateway typically provides services to a Gateway of other numbers or features and something else will say is we'll use the term DN probably if I enumerate that in a few slides or actually in another slide set but if
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1:13:05
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you hear me use the word DN this is just a acronym for Directory Number also known as an extension, okay but anyhow with H.323 both sides of the communication are very intelligent and independent of one another
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1:13:22
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So as I started to mention even as a Cisco VoIP Gateway which is typically provided by a router as it goes to communicate or even as we go to set it up in conjunction with the CUCM (Cisco Unified Communications Manager)
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1:13:45
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that Gateway that Endpoint that Router will not register with the CUCM, never see it as register okay? Now H.323 Endpoints can register to something but that is called a Gatekeeper
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1:14:01
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and CUCM cannot provide the services of being a Gatekeeper CUCM actually is nothing more than another Gateway so when CUCM as a Gateway is talking, it's a Gateway to all other phones and DN's and numbers that live behind it
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1:14:19
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when it's talking to a router Gateway, then it's communicating Gateway to Gateway and they each have their own independent dial plan hopefully the single administrator or group of administrators set-up the dial plan so that they work with each other
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1:14:42
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and that's obviously we'll take a lot more of a look at in our studies, but they do have independent dial plans and there is no registration now routers can be H.323 gatekeepers that is they can serve the role of
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1:14:58
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if you ever have and this is getting way beyond the topics of this slide deck or even this level where we're at currently but just so your aware and understanding as we move on,
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1:15:13
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if you ever do have a single router that's providing the services of both the Gateway and a gatekeeper two seperate things that a router can provide services for, it's very useful to consider them and if your drawing a Topology
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1:15:32
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in order to just conceptualize, or think about your call flows it's very useful to draw these seperate routers now they are possibly in fact one in the same router as we'll take a look at in future lessons
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1:15:48
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and demonstrations we'll see both those functionalities or services being provided in a single router but its always a good idea to draw them as logically seperate units on a piece of paper since they are logically seperate
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1:16:03
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okay so again a router can be a analog voice Gateway to IP Gateway it can be a digital PRI or T1-E1 CAS, Gateway to IP Gateway it can even be an IP to IP Gateway that is have two different call legs that are IP based
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1:16:27
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and i've used the word a couple times already in our conversation the word leg or call leg. Anytime I have any sort of a Gateway the idea is I have two call legs I have my input, or Ingress call leg and i've got my output or Egress call leg
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1:16:49
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And so maybe if a call is connected to an IP phone as well as a PSTN Public Switch Telephone Network or Central Office I might have a TDM call leg, that is the leg from the router to the actual PSTN over my digital or Analog Circuit
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1:17:16
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and then I might have my IP call leg, the call leg from the Router/Gateway to my IP phone. So for any Gateway there are always two call legs and typically when we're dealing with a call from CUCM there are 4 call legs
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1:17:36
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because there are 2 Gateways involved typically I guess if CUCM has a phone connected to another phone then there's only 2 call legs, but if I have CUCM let's say I have a phone IP phone hanging off of CUCM
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1:17:52
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there's the call leg IP from the phone to CUCM and then theres the signaling call leg from CUCM out of CUCM to the actual router voice Gateway and then if you look at the router side it has an Ingress call leg from CUCM signaling
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1:18:12
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and then an Egress to the TDM PSTN network. Okay we'll look at a lot more about call flows and call legs in a moment but I just wanted to clarify that I have used that term a few times, so again H.323 Gateways do not have to register
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1:18:31
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and are intelligent Peer-to-Peer Endpoints, however they can register to a Centralized Dial Plan Server called a gatekeeper and that's the only time they do register
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1:18:43
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and CUCM cannot be that gatekeeper so Gateways never register H.323 Gateways I should clarify never register to the CUCM. Looking at SIP or the Session Initiation Protocol, H.323 was developed by the old school ITU-T in a hundred and fifty years of service
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1:19:07
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In contrast SIP was developed by the (IETF) the people that ratify all of the standards for the internet known as RFC's so the Internet Engineering Task Force came up with SIP
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1:19:21
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And this is also a Peer-to-Peer Protocol, Endpoints are still intelligent and they still have independent dial plans so again no registration is required but also like H.323 there is an option to register
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1:19:36
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it's not something we do that often, but they certainly can register to a Centralized Dial Plan Server called a SIP proxy server and theres actually a series of servers that deal with the SIP calls
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1:19:48
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the proxy server is the one that's registered to but there are SIP redirect servers and there are others, the idea with SIP was that it's a much easier to read format in terms of the actuall messaging
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1:20:03
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when we look at Debugs and traces, SIP is actually carried in a very similar structure to as an SNTP mail header or email header. If you ever looked at those it's certainly not identical but it is similar and it's very easy to read
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1:20:22
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plain english SIP is a great protocol to work with because its so easy to debug and troubleshoot. Looking at another protocol called (MGCP) or the Media Gateway Control Protocol this is also an IETF developed protocol
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1:20:43
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and it was designed specifically by them for IP-to-PSTN voice Gateways, so really that's the niche that it serves and it does it well i'd like to say it does it well there are certainly some things that can be watched out for
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1:21:04
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and we can talk about pros and cons of MGCP versus H.323 or MGCP versus SIP at another time but it is designed just for this niche where as H.323 can be used for phone and points as well as voice Gateways SIP can be used for phone and points as well as voice Gateways
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1:21:27
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MGCP really is not meant to go on the phone, its not to say that it cannot go on the phone it cannot be a protocol that's used on the phone. I will say with CUCM or CME so of the products that Cisco offers to enterprises
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1:21:44
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it is not a protocol that is used for the phone, it is used just for the voice Gateways actual Gateways out to the PSTN or to another sort of PBX integration
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1:22:04
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Now MGCP can be run on Cisco phones however it's only used for Cisco products known as the BTS softsource something that is carrier grade and that has absolutely no place in any of our certifications
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1:22:21
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at this point any of them even on upto the CCIE course, MGCP is a client server and actually more specifically it is a Master/Slave protocol that is to say that one side always the Gateway side
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1:22:36
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is both the client and slave, its not only a client but its also a slave in a sense that it has to do exactly whatever the master you see a master tells it to do it has no ability to think independently so client endpoints are non-intelligent
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1:22:55
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the Client/Slave must register with the Server/Master and if you were given a choice between client server and Master/Slave relationship I would go with the better answer being the Master/Slave relationship
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1:23:12
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and it has to obey instructions it has no independent operation but it also has no idea of the Dial Plan so the client and the slave has no idea of the Dial Plan it doesn't need to its actually one of the better features of MGCP
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1:23:26
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there certainly are pros and cons to both MGCP as well as some of the other protocols in contrast with each other, one of MGCP's benifit is that I dont have to restate or redesign not really redesign but reconfigure and type out my Dial Plan on every single Gateway
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1:23:51
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Another and final protocol that we'll take a look at is the Skinny, or well actually SCCP or Skinny Call Control Protocol this was designed by Selsius Stysems who also built CallManager Unified Communications Manager used to be called CallManager
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1:24:07
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for those of you who are very new to voice and this was acquired by Cisco Company in 1998, so Skinny was based on the H.323 protocol from the ITU but the idea was that the H.323 Protocol was too fat
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1:24:26
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it had way too many chatty messages it was designed originally again based on H.320 which was the ISDN video conferencing protocol and wasn't really optimized for voice because yes years later there was also video overall of these VoIP
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1:24:49
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"V" didnt just stand for Voice but it stood for Video as well, however originally it just had so much information in it and the guys at Selsius said not only does it have so much information but its got a lot of
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1:25:04
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it actually doesnt have all of the necessary information we need for all of the Supplementary Services, many of the Supplementary Services that we're used by standard PBX's or standard voice Gateways PSTN Gateways
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1:25:21
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but not everything that would need to necessarily be used by endpoints such as phones and so in order to be able to develop on top of that they created they're own off chute and obviously we were able to augment and enhance it
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1:25:38
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and Skinny it down quite a lot as they saw fit so now that Cisco owns the Selsius elect property this is a Cisco propriety protocol although some of the base language and protocol information is open to public to understand and use
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1:25:55
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but not necessarily change so no one else can develop on this protocol however other PBX's other IPBX's can use this protocol asterisk as an example of a IPBX that has the ability to use Skinny inpoints such as phones
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1:26:12
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This is specifically used for Cisco devices such as IP phones and analog Gateways okay not for digital Gateways so other than IP phones that use the Skinny Control Protocol there are analog Gateways so Gateways that have a lot of FXS
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1:26:31
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actually typically just FXS and E&M voice ports, im sorry analog voice ports out two analog phones fax machines things of that nature paging systems and it can also be used for virtual type voice ports such as
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1:26:49
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voice mail ports in CUCM or Cisco Unified Communications Manager such as UNITY and UNITY connection these also use the Skinny Control Protocol
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