CCBOOTCAMP Webinar - CUBE ( Cisco Unified Border Element )

bringing now so the system is recording so we can play this back if if anyone's interested in in the future or a later date to watch and review the system so today CC boot camp we're providing a webinar on Cube cisco unified border element my name is Larry Metzger and I'm the lead instructor for unified communications with CC boot camp and my CCI number is one three nine three seven I am voice as well as route switch so I do have a pretty strong background within the CC ie world I've been working in this area for quite some time and definitely been heavily involved within the training aspects and whatnot so technical instructor consultant and again if you have questions I will be referring to the questions area once we're done with the session we will have a full Q&A so everyone will have a chance to ask questions so at this time I'll go ahead and continue on with what we're going to cover so the agenda for today is going to be just an introduction to cube what is cube what's it all about why is it important to us then we'll move into talking more in depth about sip to sip and sit h.323 with a Cisco Unified Communications Manager as well as communications manager Express we'll we will do a demonstration I've got a small set up of equipment just a demo how cube works what it does and how it functions and then we'll have a Q&A session at the very end so again you'll be able to ask your questions if you post your questions during the session I may not be able to see them just trying to keep up with all the things here but we will have the ability to review that information and talk about at the very end so just to start talking about cube functionality so cube is basically how we interconnect the void voice network so we're going to bring in the capabilities of allowing two separate voice over IP systems to talk to one another cube is also known by several different names and in the old days we used to call it an IP IP gateway or a session border controller Cisco came out with the name of unifying border element so it offers some features that may not have been available in the our operations but cube is not a whole new technology it's something that's been around for years it's in our old iOS code as an IP IP gateway or a session border controller it's available through the iOS so from that standpoint you know we implement it on Cisco IOS gateways so that would be you know your 17-yard series 1800 series 2900 3,970 200 series and I'll give you a slide that sort of lays out what the functionality and performance of each of those series would look like within your environment so the scalability of the solution it also gives us the ability to connect VoIP dial peers so we have multiple voice out peers again being able to bring in different options different protocol communications so as shown here we have h.323 to SIP h.323 to through h.323 or a sip to sip so within our our VoIP environment we do have that ability to communicate different protocols it will allow you to communicate not only different protocols but also different codecs the different codecs will require transcoding to occur but you can communicate not only different control protocols but different codecs within the VoIP communication so cube gives us a lot of functionality that just wasn't something that people were aware of because providers traditionally didn't have this ability so most companies when they were using IP to IP gateways or cube in the past we're using it because they were internally communicating different voice met works now with the service writers out there be able to give us SIP trunks into our companies or into our business that gives us a lot more features that we can utilize so it's not just an internal thing it's how we connect to the PSTN to the outside world beyond the walls of our company or organization so this is just a showing of what this looks like so either sipper h.323 it connects to our IP IP gateway so this is our gateway in the middle as you can see we've got an IP to IP gateway which could be any of the routers that support the voice capabilities and we can communicate from either side inbound VoIP diapir so as it comes into this gateway we're going to have an inbound VoIP here and with all dial peers within a gateway whether it's a regular PSTN connected gateway or an IP IP gateway we have to have an inbound and outbound dial pair those dial peers are what determined how we communicate whether it's a sip inbound to a sip outbound or an h.323 inbound to an h.323 outbound so this is going to give us that functionality being able to communicate from one side of our network to the other side of our network again just a picture depict that so you can see what's going on so why do we want queue why is this border element so important to us these days well because we have the ability to deal with our session management so on the left hand side here you can see real-time session management call it Mission Control we can we can control what calls are allowed how they get from point A to point Z where as they go through our network of course we want to put quality of service on them so QoS features gateway fallback so we have the ability for PSTN failover x' statistics and billing as well as redundancy and scalability one of the nice things about using an IP IP gateway is we're not as tied to physical hardware as we are with a say t1 PRI or other method of connecting to a PSTN where we're limited by the channels that we've got with a cube solution the scalability is much bigger within the actual routers of the gateways over on the left-hand bottom side there we look at our interworking so again looking at what can we do with this we've got a c2 3 to sip sip sip sip normalization we can deal with DTMF the dual-tone multi-frequency so your buttons as you press them how do we do DTMF be able to convert between different DTMF tones different solutions there transcoding again as I mentioned we have the ability to use g.711 2g 729 different codecs and using a transcoding implementation we would be able to fulfill the functionality between disparate systems that otherwise would not be able to talk and you can do fax and modem support so it does support those things other nice features about the demarcation so this gives us the ability to sort of create some interface between us and our carrier by using a cisco unified border element we're going to terminate those calls the real-time protocol streams the RTP stream is going to be terminated on this device which prevents the one side of the world from seeing the other side of the world so we don't have to have our carrier be able to see everything within our network we don't need to see everything within the carrier network we can hide behind this demarcation and just communicate between the endpoints so it gives us that topology topology hiding opportunity different protocol demarcations as well as our our statistics and billing from it dealing with security we have some ability to do some encryption authentication registration and really the ability to avoid toll and toll fraud so there's a lot of features within the security aspects of this solution as well that prevent the inherent unknowns within our data network so the IP network can be hacked down pretty easily but this cube will allow us to put sort of that that demarcation between us and the outside world which gives us some security and some levels of control over what's going on within our network for other reasons why we want to do this while looking at the solution it's a lower cost if we're dealing with t1 PR i--'s or other methods of connecting to our carriers we're looking at a lot of cost every time we want to add more we're adding more as far as connectivity so that PRI is costly we have to keep adding pris with cube to increase it's just a matter of scalability it's a data network connection so for the most part it's a lot cheaper these days to get data connections and then we run SIP trunks across them and that's going to be a lot cheaper than if we try to put in more and more t1 PR i--'s it's more flexible I can change how this comes into my network I can change locations that it comes into my network which brings us to the next point which is a real big one for me is this site redundancy I've had several customers where I've rolled this out for them and be able to provide them site redundancy for their inbound and outbound you know when you're dealing with t1 PR i--'s you're dealing with the ability to have D ID rollover only when that PRI connects to the same switch within your your service provider if you're trying to do site the site redundancy for D IDs you can't all you can do is roll to a single number with cube in the SIP trunks we can actually do D ID redundancy across your network have multiple ways into your network as well as allow for out of area D IDs so we're not having to deal with you know in Texas we have to have a PRI to get our D IDs for Texas or you have to pay an exorbitant amount of money to have those DIDS shipped over to another location and brought into your network with SIP it's very easy to do and very simple again the scalability is huge just being able to increase the number of lines the number of links how many calls we can handle really a big deal and ease of deployment when you're deploying the solution I'm going to show you that there's really only a handful of commands that you need to do to set it up and the functionality is very easy to deal with so here's just sort of a diagram that lay out what this looks like for you so I'll try to highlight some pieces here so we've got our site here in San Jose and we've got a call manager cluster so in the middle of this we've got our cube gateway and the cube gateway is the only thing that has to communicate with the SIP carrier so again this could be an h.323 to sip or it could be a sip to sip it doesn't have to be one the other we could change the stage street or from h.323 to sip very easily the ideas that we've got our carrier on the outside and we've got here's one location but through our IP when we are able to get a remote site still everything just comes through one sip connection so I've got a lot of customers I've worked with rolling this out again with those DIDS for the remote sites they can support those all through one inbound connection and with minimal cost and easy scalability so I don't have to have SIP trunks coming in every site like I did with t1 PR i--'s right I'd have PR i--'s coming into each and every location here I've got one connection to my carrier I can bring that in if I want to have multiple connections and create redundancy I can do that as well so I could have a connection in San Jose and say one in New York and those would be my two SIP trunk connections and then everything else basically would be remote sites that don't have to have any sort of connectivity to the carrier so we wouldn't have to have service writer connections at each and every location makes a solution very scalable very easy to manage and maintain and really a large-scale functionality continuing on just looking at what we can do with cube and how it functions looking at cubes functionality you'll see that we have the ability for media flow through which this is our standard operating procedures this is where our signaling comes from one side from one connectivity type whether it's h.323 or sip and we have signaling going to the other side again h.323 or step and in between sets our IP IP gateway now in this media flow through we've got the IP phone on the left hand side this 10.1.1.1 is going to communicate with the IP gateway the IP gateway is going to terminate that call so that call will actually end RTP stream on that gateway another call will basically be picked up between that gateway and the phone on the other side so the other side can also be IP addresses 10.1.1.1 and there's no problem with this sort of a setup and configuration because the two phones don't know about each other they have no idea what's going on or how get from point A to point C we'd be doing natin on the backside of this so the IP IP gateway obviously has to have some methodology to communicate with that different phones because obviously we can't have one device talking to two 10.1.1.1 IP addresses but we have the ability to mask that so from the phone standpoint the phones have no idea that they're the exact same address they're completely separate they are terminating their calls strictly to this IP IP gateway and that's called media flow through we also the ability to do media flow around what this does this basically means that our signaling is going to all go through on our IP IP gateway so our h.323 or sip is going to signal through but we're going to allow the media to avoid going through our IP IP gateway and actually a media established directly from one phone to the other phone this solution might be something that you would want to use within your organization dealing with a call set up if you've got numerous communications managers or different voit way providers and things of that nature you may need to control the the signaling through an IP IP gateway but you may not need to terminate RTP within it which would it give you a little more scalability as well because we aren't having to deal with the fact that this IP IP gateway is terminating a recall and generating another call until those r2p streams or transcoding we'd have the phones talking directly of course this isn't quite going to work if we're dealing with our service providers we have service providers definitely we don't want our our endpoints communicating directly across so looking at scalability within the solution if you look at the scalability everybody always asked me well how many calls or how many trunks can I support with the cubes cisco unified border element what can I do with this and the problem is that there's two factors that go into scalability on the solution actually three if we get into transcoding the factors that really emphasize here are our active calls so we have active voice calls that's our capacity of the device and then we have our calls per second so these are how many calls are we trying to handle how many calls we're trying to deal with per second so that's our set up that deals with that signaling so whether we're dealing with h.323 or sip as we bring those calls in how big and fast is the device have to be underneath in order to handle the calls that we expect so for most companies in most businesses you're looking in this range down here you're 2,800 2,900 that's reasonable you can see that we can handle 5 to 600 active calls in this range 8 to 12 calls per second of setup that's pretty reasonable for most businesses as you start to get larger and you move into the larger call call volumes you can go to the 3900 or an ASR or all the way up to the ASR thousand four which is pretty obnoxious twelve to sixteen thousand sessions I don't know of too many organizations that needs that kind of support in the single box obviously scalability on this input in multiple cube devices so you don't have to just stick with one if you go beyond the capacity of one you'd want redundancy not only in your connector e to the outside world but also in the device itself so you'd want to have multiple other devices so this just gives you kind of a feel and an idea now the third thing that I was mentioning before is the transcoding so obviously if we're e transcoding 729 to 711 codecs our transcoding resources are going to play a huge part in this scalability of the solution this is really this scalability a diagram that I've got here on screen is showing you we're talking about a standard codec on each side going through the device so Kodak filtering so we are able to do some codec filtering where we do negotiating so we negotiate a codec here you'll see that we're going to negotiate G 729 coming from the left side it's going to negotiate through the IP IP gateway and hookup G 729 on the other side so really negotiate the communication the other option we have is for the devices just directly negotiate and we put the Gateway in transparent mode and it's just going to transparently accept the codec that's being negotiated by the two end devices so it's not really involved in that negotiating from the process so that's called codec filtering on cube and it's just a choice that you make when you configure the device again on your dial peers you can change this configuration on your dial pair so that's how you'd handle which solution you want to do so at this point just to make sure that I've got people still involved with me and and not falling asleep completely I'm going to go ahead and do another poll so I just want to check in with everybody and see if everybody is awake and first off and if everybody has an idea of what do we call a cube solution what are our other names for cube IP doorway IP IP gateway voice access gateway session border controller and there's multiple answers to this solution and it looks like most people are on the ball here and I guess I'll go ahead and close that cuz I think we've got pretty much everybody answering there hang on okay yeah I got pretty much everybody answering there so I'm going to go and close that poll so of course the two answers were IP to IP gateway and session border controller I had those up on that slide moments ago on this slide there's even IP IP gateway so you should have had at least one of the two correct voice access gateway that's obviously you could call it a voice access gateway but it's not a definition of what cube is cube is session border controller or IP to IP gateway so everybody did really well on that good to see everyone's still awake out there - that's great to see I did see that we have a few questions coming in and again I'm going to be answering all questions at the at the end of the session and I'll continue on here so now I've got that closed you should still be seeing my screen and I'll just make sure of that okay so how do we get in and actually start this whole thing up well the main thing to do when you're trying to configure an IP IP gateway cube session border controller is you need to start the system to allowing that so you need to go into voice service voice and type the simple command allow connections from and - so this is really all we're looking to do we need to allow the connections and we put in which from option we want and what to option we want and again this is a unidirectional solution so what does that mean by unidirectional if I were to say from sip to h.323 then that means my inbound is allowed to be sip in my outline is allowed to be h.323 if I want to allow h.323 to sip I have to turn on that connection going the opposite direction and say allow connections h.323 to sip so we need to call out both of them so let's look at Maestri two three two h.323 or networking scenario again we can have lots of different options here but we've got a san jose site and a chicago site we got Cisco Unified Communications Manager on one side and we got communications manager Express on the other side in between we've got an IP win and on the IP win all we want to do is communicate h.323 and on the San Jose site we're going to communicate with h.323 we're going to terminate those calls on this IP d IP gateway so basically this would happen say you're partnering with the company and you want to be able to communicate between your device and their device or your organization has just acquired another company and the other company already has some pieces parts in place and you're not ready to fully integrate them into your system in fight communications manager within your your company's Network so you're in the early stage of acquisition and you're going to be integrating them in some future date at least I would hope you would that would make a lot of sense but in the meantime you want to be able to allow them to communicate with you directly but just a simple integration can be done with an IP IP gateway or queue so that's basically what this looks like and I'll continue on and show you how we would set this up so to configure it we need to first enable h.323 h.323 interworking so we're going to allow issue 2 3 2 h.323 chase tree 2 3 and then we're going to configure our dial peers because that cube device has to know how to get from one side to the other so pretty simple process here's all you have to do all right first thing we start off with is voice service voice and allow connections page 3 2 3 2 H 3 2 3 congratulations you've just turned on cue you're allowing a VoIP down here to talk to avoid I appear pretty amazing right that's all it took to command lines and we've turned on cube we have an IP to IP gateway now it actually allow this to become something bigger and better well you need dial peers we have dial peers that allow us to communicate from one side to the other so we've got to see you see em we're calling out what destination pattern goes that direction and where it goes and then we have a to see you CME destination pattern which would allow it to communicate and then the IP address of where it's going to now at the bottom there I mentioned if you want to make this sip all you have to do is add session protocol sip v2 to make it a sip table here so not a miracle there and we'll walk through that in just a moment but I just want to show here that you do have that ability to make this any time someone dials 82.0 2000 of $29.99 we're going to send it this direction whenever someone dials the three thousand two three nine nine nine that will send it to see CME we now have the ability to communicate with in our cube solution we turned on h.323 day three to three we've got a pair of h.323 dial peers and we are now functioning as a cube gateway I know everyone's looking for the miracle right that's it that's all the mojo that's the whole excitement there so looking at the next slide here we've got basically the same thing except now we're going to be talking sip so within the sip option we've got our call manager here we've got our sip carrier on the right and we've got cube in between so when we go in and we want to make this communication all we're going to do is convert at that cube gateway from AC to 3-2 sip so that way we can have a sip carry out there most companies out there and most users and engineers would probably prefer to just go ahead and run sip on their side on the communications manager so you could do a sip to sip as well really doesn't matter it's not a huge difference within the router the cube device as I showed on the other page all we do is change the session protocol the sip and we're done so I'll just go ahead and continue on and show you what this looks like so we're going to enable h.323 to sip interworking and then we're in a configuration 2 to 3 and sip dial peers and there we go so it does take a little more to allow the AC 2 3 to sip because as I mentioned last time it is unidirectional so when we say voice service void we allow h.323 to sip that's unidirectional that means our inbound can be a 2 3 and our outbound can be sip we need to allow the other solution which is sip inbound to h.323 outbound so once we've done this again we now have an h.323 to sip interworking so we are able to function and communicate it's great and then we create our dial pair so within our dial peers if we go in and create our call manager dial pair this one shows up as a an h.323 we've got the 2 dot dot dot which allows our 2020 999 and it's going to go using DTMF relay of h 2 h 2 4 5 alphanumeric so our DTMF we're going to handle in that methodology and then to an international carrier we want to call out how we're getting there so we have a 9 0 1 1 T so as our destination pattern that we're send when we see that we're going to send to the IP address located here in the ipv4 and we have a DTMF here which is RTP nte so this is what allows us to function communicate on our dial peers so we now have configured in h.323 to set continue on some things that we can do I mentioned earlier about the media flow and transparent codecs just a couple of commands as far as the dial pair goes so media flow around or flow through the default is going to be flow through so we're going to be in control and we can change that the flow around if we care - as far as codec transparent we can have our codec pass through on the dial peer and you just simply add the codec transparent on the top here and then we would just be transparently sending codec between the two endpoints so how do we configure that again looking at our diagram of our San Jose site and our Chicago site we want to make sure that we have a direct codec negotiation and that we send our direct RTP stream again in this situation we may not be trying to hide or mask who we are we're just trying to deal with the call control and the call control is going to be why we're using the IP IP gateway or the cube solution in this situation so here's the configuration the changes we added we had a couple of minor changes here and that would be the codec transparent and the media flow around to the dial peers notice again that this is only done on the dial peers we really haven't changed anything anywhere else within our cube solution we're changing on the dial Pearce's tapered dial peer configuration option within the system so since I've been talking for quite some time I think what I'm going to do is go ahead and check to see if I find any questions and see if people had anything out there so I'll try to bring up the questions all right well I'm going to turn off my showing of the screen and see if I can bring up the questions none not having any luck bring up the questions right now bear with me while I see if I find the questions here there we go you bear with me there's just a few questions here I'm clicking through and some are are just responding to me and you're not really question so is it SBC required with SIP trunk service so if you're getting a sip trunk service from your service provider then a cube or an SBC a session border controller cube IP IP gateway typically is required the main reason for that is you wouldn't want to directly connect say your communications manager and all your IP phones on the back side into your service writer because remember RTP streams must be from endpoint endpoint so if you're going to be connecting to a a SIP trunk with a service provider typically you don't want that service provider knowing all of your internal network and being able to see your internal network even if it's just your voice network so you would use a cube solution and that's that's the key to why this has become such a big ordeal recently the reason it's become such a big ordeal is because many of us are finding that the carriers are providing SIP trunks for great price for a solution that really gives us some performance and so we're looking at going to SIP trunk so cube is become a key issue because of that will the slides be available for download no we are recording the session so you will be able to go back and review the session you might be able to get to the slides I might be able to share them with you after the session but they're not easily downloadable from the site and I can see what what availability we can make for those if people are interested another question is do you need DSPs for cube DSPs are not required for cube the configurations I've shown so far do not require them that's because I'm communicating the same protocol G 729 on each side or g.711 on each side so the codec is the same going across if I were to try and do g.711 on one side of my communication in G 729 on the other I would need a transcoding which case DSPs are required so they're not required inherently but if you're trying to actually between different codecs you will have to have them okay there was question about the 500 600 call capacity media flow through or flow around that would be with media flow through obviously media flow around would require less resources because we're strictly dealing with a call setup alright with that I'm going to go ahead and go back to sharing what's on my screen here and I'll continue with some more questions later but I just thought I'd answer some because I noticed that there were several building up I didn't want to leave anybody hanging so I'll come back and answer more at the end of the session so this just shows us our verification plan yep there we go how do we verify what's going on within our solution and within the cube environment so there are some key commands that we have in here and those are the show sip call this will allow us within our device whether we're on the call manager Express or what have you or on the cube itself we can show the sip call so this will allow us to see what calls are set up we can show h.323 gateway H 2 to 5 is going to show our call set ups so we can see what has been set up by the device or our cube gateway is going to show information here as well as just show HTTP gateway show voice call status now on on the cube device you won't see anything with this when you do a show voice call status you're going to get nothing on because it's not dealing with initiating a call if you're on a cme device you will see the call status when calls are placed so I'll show this command on both so you can see how that looks within our lab environment that I'm going to show you in just a moment the one that's really helpful is this show voice RTP connections you can use the details option there to show a deeper information and to give you more details about the solution this works really well for whether you're on the CME gateway or you're on the actual cube gateway you'll be able to see exactly what RTP streams are being handled by the device so that's a great command to show what's going on show call active voice so this will show you what calls are active on the device and we'll see this again both on on the cube and on the CME solution and then the debug dial pair so if you want to see what dial peers are actually being hit by the calls you can see your inbound and outbound dial peers you and do the debug die up your voice all if you want to see all the information it floods your gateway pretty hard so the in/out usually is a better solution but if you want to see all the details and really dig into why am I having a problem typically I find that if you're not getting your calls going across properly the debug on the dial peers is a huge help because unless you've been around h.323 and sip configurations for a long time most engineers don't fully understand the HDTV dial peers I mean you do this debug you can see which dial peer was hit on the inbound which Gallup here was hit on the outbound and you can truly understand what's going on within the gateway so you can you can figure out where my problem is most of the time the problem is that you haven't controlled the call well enough I'm actually as part of the CC boot camp program I provide out some monthly articles for our newsletter and I've been writing some articles regarding h.323 dial peers because I find that most engineers don't fully understand dial Pearson and it's something that the more you understand about it the better off you'll be especially for tube cube absolutely you need to understand how your Dow peers are functioning how do I get in how do I get out where is my call going and you absolutely like everything within our network we want to control it so we want to be in control of how it comes into us and how it goes out so I'm just going to show you what our demo network looks like so this is basically our corporate office it's a Cisco Unified Communications Manager Express and I use see of me because a little easier to show some of the configurations on it and how I set up the dial peers on it rather than following showing a communications manager where basically you see an h.323 IP address or a sip IP address and not much else and also it was a really fast and easy for me to set up in a demo mode for you then I'll have on here our cube router so this would be our IP to IP gateway and it's going to communicate across over into our theoretical SIP carrier so I've got the setup is both h.323 and sip depending on which phone number your dial so I've got both h.323 and sip configurations on that cm II device and I'll show you how those work and I'll show you how the configuration is set up within the cube so you can see how that functions another thing I'm gonna show you is the fact that I've got this phone over here 2001 in an IP address that this SIP carrier which is also Communications Manager Express OCME device in my network for demo purposes it represents the PSTN the SIP device has no idea about the IP address of this phone so obviously there's no way I could do a pass-through solution I wanted to show that cube has to terminate these calls my RTP stream has to be terminated on the cube device because the SIP carrier has no idea about this phone and that's the way you'd want it in in your network typically as well we don't want to open up our networks and show and give access to everything within our network to that provider so this is the demo network and what it looks like and now I believe the best option here is rock slide now I need to break out and go into the actual demo portion so within the demo portion here's my CME device so as you can see I've got a CME device running on my network and that device right now has a phone plugged in and that phone number on that phone is the 408 5i5 2001 with the extension actually being 2001 and I'm going to go ahead and bring this up and I'm going to dial a phone number and you can hear that I'm getting a failed call that's because I haven't configured kube yet so at this point I can do a show he phones on this device you can see that I do have a phone hooked into this it's at address of 192 168 1 2020 it's a Cisco 79 61 device and I can do a show IP route so you can see on here I've got the 120 Network the dot one and the dot - so 192 168 dot one one nine two one six 8.2 192.168.1 20 I have no knowledge of the 192.168.1 is configuration so cme device can't see the other side which is exactly what we wanted we want to make sure that they can't see what's going on on the cube device i do a show IP route and what you'll see here is it does know about the dot one the dot 2.3 as well as the one twenty it has to know about the one twenty network because it is going to terminate the call going to that phone so it has to know about that phone so the cube device has to know about my network on my side that's know about whatever it's going to talk to on the other side but it is the demarcation point between me and the other side of the world so as you can see it does and I'm just going to do a show run and show you that this is just a basic router right now it has an IP address there's nothing configured on it all we've got some interfaces there are some t1 cards in them but they're not lit up they're just t1 sitting there for lab demo of other nature as you can see one static route no other configurations no dial pair so there's nothing up my sleeve now I'm going to show you what the PSTN looks like so I'm going to a show IP route on this side and you'll see that it only knows about the dot one the dot 3 so 191 6 8.1 and 191 6 8.3 the dot 1 is really my network so that way I can tell that into these devices so that's my back back-end connection to all the devices so that's why the dot 1 is on all of them but really the dot 3 the dot 2 are the networks that we're trying to communicate and then the 120 is my VoIP phones so again you can see what's there now I'm going to show you what the configuration changes I'm going to add in our so the first one voice service avoids so we're going to go into the VoIP configuration and we're going to turn on connections I'm going to allow all the connections so I'm going to H 3 2 3 2 3 2 3 I'm going to allow HT 2 3 to sip sip - 3 2 3 & sip - sip so I'm just going to turn on all the different connection options we have down below this I have some housekeeping and that's the voice translation rules I need this because on my void Donnell appears they don't automatically strict digits so when I set up my outbound dialing rule which I set up a really basic one not what I would do personally in the lab or in the real world but for lab demo it works really well but for the real world I would never do it so this voice translation rule is is really there to strip off the nine one so when my user dial is a nine one plus the ten digits this is just going to strip off the nine the one and send the ten digits to the carrier and you can see I call it PSTN out so it's going to be just a profile now I'm translating the called phone number so the voice translation rule in the voice translation profile are really housekeeping those aren't part of a cube configuration but because you may need to do digit manipulation I include them here so you'd see this is what I'm doing within my network so you have an understanding of why they're there down below it I have my dial pure voice 100 voice and this dial pier is an incoming call dial pier so this is my inbound dial pier and I did this separately to show you what's going on to make it so I have inbound and an outbound to keep it clear so 100 is basically begins with a nine dot T so anything that comes in from the CM e side which is going to have a nine as a prefix so when i dial a nine one or if i dialed a nine whatever i would accept that into my cube device the only thing I'm doing on this is I'm setting DTMF relay on here H 2 2 4 5 alpha and I've got no VAD so I'm not doing voice activated detection so I turn off bad because we never like to use bad real-world or lab turn off bad bad is bad next to here because I want to do both the SIP and h.323 I was more specific on this dial P R so down P R 1 10 I have a destination pattern of begins with 9 1 1 and ends something very very specific only when someone dials 911 1 will this now appear be used this is an inbound dial peer at this point the session protocol I call out next so this is a sip dial peer so I now have an inbound sip because I'm going to also show you that you can do this all in one I have a session target because on the session target I'm going to go out to ipv4 192 168 3.2 which is my gateway on the other side so this is my PSTN side the incoming called again this is how we bring it in destination pairing goes out and here I've got DTMF relay of RTP n to e which is what we typically would use for a sip endpoint so we're going to communicate sip and then no bad again no bad we always turn bad off so you can see I've got one down here where I used it as strictly an incoming called the other one I've got destination pattern and an incoming called so it's both an in and and out so 110 would be used as an inbound as well as an outbound inbound is turned by the incoming called number the outbound is by destination pattern and then an ipv4 session target that's how I get my outbound so the next one down doing down here voice 200 this one has the destination pattern of four oh eight five five five $2.00 so this is my incoming from the PSTN so this would be my incoming from my service writer so my outside world where my carrier is Verizon AT&T any one of the number of companies out there today that offers SIP trunking to you this would be my inbound from the outside world or be my outbound from the outside world it's going to go to my communications manager express in this case which is 192.168.20.10 outbound towards my c CU cme now my voice 1000 this is going to be my outgoing to the PSTN then putting my translation profile on remember i want to strip off that nine one so I'm putting that on here I'm saying the destination pattern is nine it's a sip and it's going over to the PSTN and then here's my inbound coming in coming called from my carrier my service provider it is two thousand so this is my end from my carrier my now it's my carrier now I've got my out to my communication manager express my in-and-out for nine one one in my inbound from CME going back to the cube device so that's all the configs it's going to take to make this work and it's real simple I'm just going to copy all of them just do a ctrl C on that I'm going to go out bring up my cube device so on screen we've got the cube device now and I'm going to do a config T paste those configs do an end and now I can do a show run and now you'll see that I've got my voice translation rule in place I've got my voice service VoIP with my allow connections at the top there that gets me through my cube so this is what sets up my IP IP gateway I'm allowing those connections I've got my dial pair showing up here at the bottom 100 110 200 1000 2000 we now have the ability to place calls so I'm just going to write them on this real quick hate to lose config so always do a right now moments ago when I first dialed you heard when I pressed the 9-1-1 it didn't ring now we have rain and we have established a phone call through the cube so now that I have the the phone call established through Q I'm going to do some of those show commands I mentioned earlier so some of the things that I threw out earlier we're a show sip call so I sip call you and see here that I've got a sip call in place and this call is state active calling number is 408 5 by 5 2001 so that's my calling and number so it was originated on my CM e side my call number is 9 1 1 so you can see here that I've got my call out to 9 1 1 my source IP address where it started off where it went to stream is active shows my codec DTMF RTP nte so everything's here next piece again the other side of the call because this is a two-step process sip on both sides one side and the other side so we have both sides of the call showing up here again from 408 five five five two thousand one two nine one one so you can see that that shows up there's my show sip call so that's one way to see what's going on with the call show h.323 gateway h2 to 5 you can see that I have set ups if I had some call setup in process I've placed some calls on here these were done previously because obviously I'm in used Nations recall on this call it was just a sip call but I did this earlier show h.323 gateway and this just gives me more details this is going to give me all the information about what was going on within the call so you'll get a lot more information if you just do the show h.323 gateway command you'll see that other things that we can do here again show voice call status I mentioned you would not see any call status on the cube device there it is you won't see it under a show voice RTP connections and you'll see that I've got two connections so this is basically showing again the call legs call ID destination call call it a destination ecology so I have two active RTP connections one going from sip to my see me on my side one going over to my PSTN so there's my two calls now the other thing that I can do with this is I can do a detail and here you'll see that gives me a little more information than I got before and I get the detail it does show me some detail about who called in calling information so my calling your information shows here call Ling and who did I call and I'm one see a little more information when you do that detail on here the other thing that we can also see as the show call active voice and this one gets pretty lengthy so I'm going to just scroll through it real quick because I can do the same thing with brief and the brief gives me a lot less information so it shows sip call legs to h.323 call legs zero call agent controlled call legs gives me details about the call when it was originated who originated it who called there's a 9-1-1 showing up there so i know that's called went to 9-1-1 there's a lot of information here more information if you do it without the brief city and see a lot about what's going on I'm also going to bring up the CMEs this is my cm II device and I'm going to do a show sip call on this device what you'll see on this one is that I have one so I'm going to scroll up there I have one sip call leg why because this is the CMEs the originator so it's originating sip call if we do a show I won't bother with the h.323 because we're not doing it right now voice call status you'll see on this one I do get an active call because this is a CMU device and it has its holding or maintaining controlling that active call for my telephone here on the CME's side so you will see information on that one I'm going to do the other options the show voice RTP connections and this one shows one RTP connection so you'll see that it's has one half of the conversation so those are the commands that you'll be able to do on your side and be able to see different things happening so I'm going to hang up the SIP call now I'm going to bring up a an H 3 2 3 call next so this time what we're going to do is I'm going to go ahead and dial the other phone which is a nine one nine seven two five five five one two one two so now this call should be an H 3 2 3 2 sip so I'm going to go back over to my cube first and now the cube device I'm going to do the same commands I'm going to do a show sip call if you'll see that I've got the sip call active and going I'm going to a show h.323 gateway H 2 2 5 and you'll see my calls and facilitation of H 2 2 3 calls again I'll do the show voice call status which again gives me nothing I'm not going to get any call status because I'm not in control of to call the calls are being controlled by the CM e on the ends I'm going to do a show voice RTP connections I still see that I've got to call connections so that's why I know I have the two calls happening now I do the same thing with the detail now you can see what the calls are so it gives me more details about the call so the number that I called nine one nine seven two five two five one two one two who is the caller the 408 five two five two thousand one so I can see the calls and where they're going how they're being handled who's in control them it's the RTP function and then I've got the show call active voice and again I can do this with brief just trying to see what the call legs look like now here what you'll like about this one is again I can see my sip call at the top telephone call legs I have a sip call and an h.323 call so it tells me what I'm actually dealing with I have to call legs ones running sip ones running h.323 pretty cool stuff right let's go back over to the CM e and I can do again Taizo commands show sip call and i'll get nothing on a street on the this side because I'm running h.323 currently so there there's no ability to see the sip goals because I didn't initiate a sip call I use the h.323 dial peer if I do the show h.323 gateway i'll get details about the setup of the h.323 calls so I can see all the details about calls that were sent received and how they function within the system so it gives you that ability to see what's going on within your calls again I can continue on with the show voice call status actually that I have a call active which is good I should if I do a show voice RTP connections I see that I've got one connection again I'm on the CM e side so I should only see one because I originated this one and when I do the detail of it I get more information again as you're working through and troubleshooting be able to get detailed helps so one things I always recommend is if you're not sure of the commands just hit a question mark at the end so when you do the show wipe wipe RTP connections unit question mark and it will tell you that you can do detail which gives you a little more information than you have before which is a good thing and then the show call active voice brief and it shows me my call legs again I have one h.323 call light so that's the calls going back and forth between them the other thing I want to do is on the cube I'm going to go ahead and make sure I turn on monitor turned on and then we do the debug so I'm going to do a debug dial peer voice in out well helps why spell it right debug my bad voice dial peer turn on debugging so I can watch those dial peers as they come in and out I'm going to try that call again now you can see with the debug what happened on that I can see the call number so I have calling number shows up real quick right here so I have my calling number showing up on screen the 408 505 2001 who did I call nine one nine seven two five a five one two one two where did that where'd that go what did I do with it well if I look down below I keep looking down through the system I see results success down here was dollar 100 so that's my incoming dial pair so it matched on incoming and it says Dow pier 100 is where it matched to then when we look down we should see that we went out bouncer down underneath as we keep looking down we'll find at the very bottom we have our matched outbound dial pier which was 1,000 so that's how we got from point A to point Z using debugs and I'm going to turn off debugging because I don't like to leave it on being a good engineer I try to avoid leaving those things on on my system so that's basically the extent of what I wanted to show today in regards to doing the demo just showing that you know I can establish calls through that cube the cube does terminate the calls the RTP streams you saw that with my show of the the cipa so as well as the show void RTP connections so at this point what I want to do is open it up to the questions and I'm going to try to go through and find all the questions and make sure I've answered everybody's questions within the system and it looks like we have tons of them so what I want to do is just see if I can open that up to two questions bear with me as I look through the list for everybody's questions I'm going to stop showing my screen so that way I can bring up the question and answers here okay well it appeared I'm having some trouble getting the questions section to pop up oh there it is getting used to the the interface here haven't used that GoToMeeting in a little while so are there enhancements for monitoring status of active connections above the usual show call active voice obviously you can you can do the debug on the dial peer to see that what's going on within dial peers as calls come in and go out I've used that to you know figure out what's going on with calls or problems within the system as far as monitoring the status of active connections the show commands are typically the the only way to monitor those unless you're using third-party applications or one of Cisco's monitoring tools for that built into the actual iOS there's no easy way to do it it would really be off of the the show commands that you'd be finding information and be able to view that and a question just about be able to view this later yes you will build a view it later I am recording the session and that will get posted up on the system so you will be able to do once I've posted the the recording you will be able to come in and watch that at a later time let's see question just regarding you know for the CCIE lab exam without you know breaching NDA what types of questions could they ask us on cube similar to what I just did today it's a really as far as what can you be asked to do on cube obviously you want to know how to set up cube which is pretty simple and for safety sake I of course would set up all four options you know you're a tree to three days 2 to 3 H 3 to 3 to sip sip to h.323 and sip to sip is all your connection options so you're not to worry you've got them all configured the difficulty comes in just as everybody knows on the CCIE voice exam it's not the idea of the question per se being difficult it's the four hundred little minor details that they will not mention to you that you have to figure out for yourself such as asking you to change your caller ID calling caller ID meaning your a and I so having to manipulate your an e within the call having to manipulate the deenis or the fun dimension called - having to manipulate a call to use PSTN is back up to your cube solution so if your cube call were to fail backing that up with PSTN DSPs the they could ask you to make sure that the local call where g.711 but your carrier is only accepting G 729 so f nod to transcode with it and again that's that's not difficult first off you just have to have DSPs on your device configure your DSPs as transcoders and now the fun part setup telephone service because there's no way to register DSPs to cube in and of itself so you have to actually make that cube gateway act like cm e so you're next on telephone service and maybe act like see me and register you register your DSPs - CM e itself that will allow it to invoke transcoding when it needs it for cube so really the the basics they're going to ask you to do some some basic cube configuration and what could they ask you to get more difficult well think about it I mean there's a lots of different little pieces media flow through or flow around transparency and the codec selection so there's some things of that nature they can ask to trip you up as usual so I don't believe I've reached any NDA there especially since you know I haven't taken the current version of the exam I passed my CCIE on the previous version could you explain DTMF relay RTP ntp II digit drop in H 2 H 2 4 5 again DTMF relay are the RTP nte option on there that's what we typically use for sip that is a standard option that cisco call manager uses first sip and we would want to use whatever matches our service provider so our carrier typically your carrier will tell you what DTMF relay they want used and you'll use that the digit drop means that if the system were to see that command coming in if it's if you've configured it with digit drop if a digit were actually in the RTP stream it would drop the digit in the stream because it's going to do it using RTP in NTE the last piece being h2 for 5 means that we're going to allow it to negotiate one or the other so you can set it up to with multiples on that line if the system sees multiple is on the line then it will negotiate to whatever the other side wants so DTMF relay gives you some optional choices again the digit drop is just saying that if it sees it in stream it's going to drop the digit so it's not seen as part of the stream can a firewall be used instead of SVC you can put a firewall in place you can run the iOS firewall on the same device you can run a picture and a sa firewall in place some third-party firewall the key is you have to open up ports now the good news and the bad news your sip ports are pretty well known the bad news is your RTP ports are a huge range so you have to open up the huge UDP range for RTP because you've got to communicate RTP across this which isn't such a bad ordeal because again you're not sharing your routing with the other side so as far as your carrier goes your shirt's writer it's pretty easy you can say anything from this IP to this IP address within this UDP range allow and it's not a big ordeal because really it's between the two IP addresses your cube device to their sip device and that's it because that's the only thing you're ever going to talk to and the only thing they're ever going to talk to your cube is what allows them to talk to everything else again it's it's terminating the RTP stream so you can make this very safe and secure and you can use matting with the solution as well you had a single IP on the cube box on the drawings I may have had a single IP obviously the IP address on the cube box would be for both directions so typically you'd have an IP on it on the each side of the cube so one going to your carrier one going to your internal and you don't share your internal with the carrier so the carrier has no idea what's going on what's inside your network most carriers provide you with an outside IP address the one thing that Cisco does mention about using cube is that you can utilize cube with an internet connection so your service rider can provide you your internet and a SIP trunk on the same connection so say you get a a 5 mega from your carrier you could do that personally I'm against that cuz I would prefer that we we keep our our SIP trunk on a separate sip connection that it's a data connection strictly for sip strictly for phone calls my QoS on it is very clean my carrier is very clean on it and my security on it becomes much easier but I'm not trying to secure the internet I'm just dealing with a SIP trunk where I have known sip device on the other side so that's that's how I would've view that this cube allow for authentication there is some authentication methods to it that was shown on one of the slides authentication is an option within cube gatekeeper cube configuration I won't have time to go over gatekeeper and cube that's the via zones so within cute within gatekeeper you can use via zones and the via zone can go off to cube and terminate the streams there that topic would take me several hours to go over and really dig into to demonstrate and give you more details on it that could be something I can do at a later date and time where I just focus on the via since we've covered you as part of this I wouldn't cover cube again I would just deal with the gatekeeper cube configuration so nationality thank you for that option Baba I will use that as a future one of these webinars and go over that in the future is that that's a perfect topic and this cube topic leads right into that gatekeeper invite out via communication conversation so that'll work really well 26:21 XM way I can use the the 26:21 does support it my cube gateway today I didn't show you all what my cube was today my cube for today was actually a 1700 series router why do I use a 17 yard series router well that's what I happen to have in my my home lab I've got a pretty healthy lab of equipment but I don't have you know twenty nine hundred and thirty nine hours laying around so I use a 17 hard series they perform obviously less capabilities less performance if you're looking for CCI you the lab practice a 1700 series router will do it a 2600 series router will do it just have to make sure you've got the right iOS version on there and the right iOS feature set so you have the feature set that will allow you to run it you can also practice cubes for those you who use Dynamix you can use Dynamix practice cube as well again it will it will do everything that does not require a DSP so as long as you don't need DSPs you can run it in Dynamix will the second session target will be two ISP address or internal address obviously we have an inbound and outbound dial peers so the outbound dial peers always have to point to the device that you're trying to get to so you'd have an inbound dial pair that would just receiver accept the inbound call from the service writer you can reuse and have the same Dow peer be your outbound dial peer so it's up to you how you want to your down peers on that as far as the session target yes the session target on your outbound going to the service rider would be the IP address that they provide to you do you need a pack key and licensing if you're using the 2130 900 series they do require licensing on them the 15 dot X code does I know Cisco is backed away from some of the packs I'm not sure if cubes still requires it I know some of the other iOS feature sets cisco backed away from requiring the license keys on them the last I read on cube was it still required a license but that may have changed or may be changing but for the most part the way they were working was definitely on the 15 dot X code it was requiring packs and license keys so you do need to get your packs and licenses like that's one that you just need to double check because I know Cisco did back back away from that they were having some some issues with customers and whatnot having all the service with licenses and the pack keys for feature sets on those newer 2930 915 dot X code versions why incoming and destination pattern on the same dial pere the way I was doing it I was just showing that you have the option to do it on both it's not necessary that you do some people like to keep it straight in their head so they keep them separate so they do in inbound peers and outbound peers just so it stays straight it's up to you it's a personal preference you don't have to you can the thing to remember and one of the reasons I showed it separately is to remind people that the session target with an ipv4 has nothing to do with the inbound some people get confused and think that it's going to match by the source IP address of who's initiating the HTTP conversation that has nothing to do with it I've got one of my articles my first article I go over exactly how we select our inbound dial peers and the incoming called is the first thing that it matches if there's a match to incoming called it's going to use it so that's the first option destination pattern will match the ante on the next option so on and so forth but you know inbound and outbound it's up to you how you want a program it's just a personal choice call be coming from inside the network to divider with pattern 9-1-1 but not likely to see incoming pattern now I'm from PSTN to inside so curious on this dial pair config let me see if I can show my screen again and I'll bring up my my config here on the dial Pierre config I have a dial Pierre nine one one and this one is an inbound from CME and an outbound to the service Rider so that's why I have an inbound and outbound on CM e I would only have one CM II would have a single outbound dial Pierre but because this is on cube cube does have an inbound 911 one its inbound coming from CME or call manager if you were doing an h.323 or a SIP trunk to call manager the cube would have an inbound 9-1-1 coming from your internal network and it would have an outbound 9-1-1 going to the service rider so i have these just happen to be I combined the two together so that's why I have that just another question in regards to why I combine dial peers again on why I combine dial pairs it was just to show that you can not really any other reason than that just to show that it is something you can do not that you need to or have to any special consideration in case there's a firewall between the cube and the service writer the only special considerations that you really have deal with ports making sure that you have your ports open such as you can establish RTP the biggest issue is everybody of course will think about their SIP ports and they'll make sure that their sip is open so the SIP can get through the bigger issue is your real-time protocol your RTP streams because that's a huge range of UDP ports those need to be open to allow that communication as well so those are the two two aspects that you have to think about when you do put a firewall in the network the DTMF that's most common for sip that's going to be your RTP nte that is your standard for call manager it uses RTP nte if you're dealing with CU e e cisco unity express it requires sip notify so Cu II is sip notify as far as the rest of the world typically RTP NT e a standard DTMF for h.323 is h2 for five alpha numeric so DTMF relay for h.323 will be H two four five - alpha numeric so really if you're looking at what I showed today I used in my demo I use the standards of the industry what most people will use and definitely you'll want to use unless someone tells you otherwise if you're told different by your carrier use whatever they say if they say nothing then use the RTP n - e for sip and H two for five alphanumeric for h.323 as far as there's a question here will you be conducting the mock bootcamp also I am the lead instructor for voice and I will be teaching pretty much all of the CCIE voice level courses for CC boot camp not to say that there isn't potential that I'll be teaching one class and we'll have another instructor teaching this those classes but typically the classes that CC boot camp offers I will be teaching again I try to bring in some of the real world as well as lab environment into the classroom so I hope that you all appreciate those opportunities after the class the video there's a question about how will you access the video of this session that will be posted online so you'll see that there all right with that I'm going to

Channel: ccietraining

Views: 30,081

Rating: 4.90625 out of 5

Keywords: cube, Cisco, Voice, CCIE, Unified, Border, Element, Webinar, CCBOOTCAMP, Training, CCNP, CCNA, Network, certification, training, ipexpert, internetwork, expert

Id: 5DjYdxnleGo

Length: 75min 28sec (4528 seconds)

Published: Wed Aug 10 2011