Configuring and Managing Direct Routing in Microsoft Teams

Configuring and Managing Direct Routing in Microsoft Teams

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welcome to our team's IT Pro Academy training this is a multi-part training on direct running in this lesson we will be covering configuring and managing director adding with Microsoft teams in addition to this training you can find other IT Pro Academy trainings at our AKMs forward slash teams academy link if you have feedback that you'd like to provide on this session you can see the links and instructions noted on the slide here if you have feedback on the product itself make sure that you follow our AKMs slash team's feedback
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link you can see at the bottom of the slide the current version of this presentation now some of the key learnings that we're going to cover in this session direct rally allows you to connect your existing a phone infrastructure to the Microsoft team service director adding itself will require a certified session border controller now we'll cover what those are as we go throughout the presentation we will also cover the configuration for direct routing in detail during the session but we do also have it clearly documented on our Doc's
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portal let's take a look at the full agenda for the multi-part training on direct routing the first lesson will cover what is direct routing when to use it and how to properly plan for direct routing the second lesson will cover configuring direct routing and managing director adding and the third lesson will cover migrating to direct routing the lesson that you're currently taking is highlighted on this slide let's begin this lesson by talking about configuration of direct routing the
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configuration steps for direct routing are outlined on the slide starting on the left you'll see that we'll begin by configuring the session border controller itself once we've properly configured the SVC we will in connect or register the SVC with the online service from there we can configure the voice routing so that we understand how to properly route calls to the public switch telephone network and then optionally if necessary we can configure the translation of phone numbers finally our last step will be to enable the
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users for direct routing when configuring the session border controller itself well want to ensure that we follow the configuration documentation that is provided by the specific SBC vendors if you follow the AKMs link on this slide that will take you to a landing page where you can jump to the respective documentation for the specific SBC vendor of choice at a high level the configuration of the SBC will include items such as configuration yes SPC license a configuration of the LAN
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and LAN IP interfaces configuration of the certificate used on the device itself the configuration of the ports used for both signaling and media the configuration of these sip options as well as secure RTP which is encrypted media the codecs that will be used both between the session border controller and teams as well as between the session border controller and whatever other telephony component you're interconnecting and finally be routing that is necessary to route the calls
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from the team's environment to the PSTN from the PSTN to teams and to many other telephony components that you are interconnecting with the session border controller on the service side we connect or register the session border controller using a PowerShell command lit do - CS online PSTN gateway this will allow the service to realize the fqdn of the session border controller the port that the session border controller will be
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listening on the maximum concurrent sessions that the SVC is capable of handling and whether the session border controller is enabled or not the parameters that I'm showing here are some of the basic minimum parameters that we need to set you can see by running this command lit I've registered the SVC by name so this could be SVC one contoso com for example the port that my SBC is listening on is fifty sixty eight I have set the max concurrent sessions to fifty this is not a requirement but
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it's a really good practice to get into because you can align this together with the maximum call limit that the SBC itself can configure that way the system will know if there are too many calls we don't send anymore calls to that social border controller because it's at capacity and then of course enable true you could set this enabled false if you were taking it out for maintenance or things of that nature there are some other optional settings that you see here that I'll cover on the remaining slides but you can see this is the basic commandment that we need to run to register the session border controller
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with the service now remember the domain portion that we use for the SBC must match one of the registered domains one of the vanity domains we have in the tenant remember that you cannot use the dot on microsoft.com domain name as we talked about in the planning module this slide in the next slide some of the more common parameters that we will use with the cs on line PSTN gateway tenant remote powershell commandlets this is not an exhaustive list of parameters this is just some of
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the more common ones that we will use when we configure the session border controller in the service we have to configure the sip signaling port right this is to let us know what port the session border controller is listening on for the sip connection for the service to be able to make the SIP connection into of course we need to tell whether this is enabled or not these two items forward Pai and forward call history forward Pai indicates whether we use the piya certain identity
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which is what Pai stands for whether this header will be forwarded along with the call this Pai header is important for providing a way to verify the identity of the caller the default for this will be false so if you want to use this you need to make sure you explicitly set it to true likewise you can use this forward call history parameter which will indicate whether call history information is forwarded through the trunk if in fact we do set this to true the SIP proxy that we use in the service will actually send two
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additional headers the history info and the referred by headers again if you want to use this you need to explicitly set this to be true because the default is false some other common parameters that we use sip options enabled this should always be true and it is true by default because the SIP options is a polling mechanism that we use to ensure that the Gateway is still alive and well if we turn off sip options we don't really have a way to monitor and alert whether the SPC has gone offline or not so
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always a good idea to keep these enable Maxo concurrent sessions another parameter that can be used by the alerting system and is great to align this to the maximum sessions that you configure on the session border controller that way if the number of concurrent sessions is 90% or higher than whatever this value is you'll get an alert in the admin console to let you know that this gateway is nearing capacity you can set an able fast failover timer when this is true outbound calls that aren't answered by the Gateway within a 10-second interval
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can get routed to the next available trunk and if there aren't any additional trunks well then will automatically drop the call this is useful in an organization or we maybe have slow networks or we have slow gateway responses this could result in a call getting dropped unnecessarily the default value for this is true if you need to you can set this to false and then there's be a bypass as we discussed in the optimizing section for planning if we wanted to enable media bypass for this particular gateway we would simply
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set the media bypass parameter to true now that we've talked about registering the session border controller with the service let's step this all the way through from creating the Gateway to creating a simple voice routing configuration so as we talked about on the last slide our first mechanism is to ensure that we register the session border controller with the service and here you can see I'm using new CSS online PSTN gateway to register at this session border controller with a sip signaling port of
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5068 to the service as we talked about in our planning in order for me to be able to set up my voice routing I need my combination of usages voice routing policies and routes so in this case since this is a very basic configuration I'm going to create a new usage sets es online PSTN usage identity of global and then I'm gonna add this new usage name and I'm simply going to call this usage unrestricted I will then associate that
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usage with a new voice route so here you see I do my new CS online voice route the name of the route I'm also going to call unrestricted and you'll see my number pattern here is dot star that's basically a catch-all it's gonna match any number pattern my online PSTN gateway list is my session border controller SBC one Koto's com its priority one because it's the only route that I have and here I've associated the usage online PSTN usage that I created earlier to this voice route right so
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there's my usage of unrestricted there's my route of unrestricted with this gateway matching basically any going to this session border controller the last piece of the puzzle is to create the voice routing policy so I do new CS online voice routing policy in this case I also call this unrestricted - online PSTN usages this is what links the PSTN usage of unrestricted to this voice routing policy and that's what we've done right here so now my usage is
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linked to the voice routing policy my usage is then linked to the route my route is then linked to the session border controller and we are ready to roll let's look at a slightly more advanced routing configuration here much like we saw in the previous one we're gonna start by registering the online PSTN gateway in this case because this is a little more advanced I want to have some additional redundancy so I'm gonna build multiple session border controllers into the service so here we go there's my
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Commandments for SBC 1 and SBC - I'm gonna create a PSTN usage this time this usage is specifically for making calls to the US and Canada now again this sets es online PSTN usage by itself really doesn't do anything because you notice this usage is really tied to anything it's not tied to routing policy it's not tied to a user it's not tied to Gateway nothing right I'm gonna build out my voice routes my first voice route I'm going to do specific 2 + 1 45 + + 1 206
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these are my Redmon numbers were my seattle numbers there's my QC s on online voice route I call it Redmon one my number patterns that I'm gonna match my gateway list priority and then map that to this usage as you see individual on the left I have another pair of SBC's three and four that I added into the service and I have a new set of routes you see us online voice route this happens to be Redmond - now you notice this has the same number
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pattern that we see for this first route but notice what's different different set of gateways and different priority all right so now this Redmond tear out this is a backup route as long as these session border controllers are available this route has the higher priority this will always be used if both of these session border controllers go down this would not be a valid route we would fall to this route and I would send the calls out through SBC's 3 & 4 so now I have a
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little bit of redundancy two more session border controllers in the mix I've added five and I've added six and hears my voice route for the remaining calls UCS online voice route identity other plus so this is basically anything that starts with plus one as you can see by this number pattern it's a full 11 digit dial but it starts with plus one my gateways are SBC five and six and again I've linked it to the same usage of US and Canada so now my usages are linked
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to my respective routes which are linked to their respective gateways now I can build my voice routing policy new CS online voice routing policy and I can associate this US and Canada usage with this voice routing policy and away we go now again at this point all we've done is configure all of the routing we haven't yet assigned it to the user we're going to do that as we go through the next couple of steps and we get a chance to actually enable the user for
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direct routing number translation is an optional step that we could use it may be necessary for interrupts with various session border controllers in specific scenarios now the number translation rules are applied at the gateway or at the SBC level and they can be applied in both directions meaning they can be applied in Bound that would be a call that comes in from a PSTN endpoint to a team's client or they could be applied for outbound calls that would be a call that
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starts with a team's client and goes to a PSTN endpoint all the number translation is accomplished using tenant remote power shell or TRPs for short in tenant remote PowerShell we create and manage these rules with the CS team's translation rule command line and you have various verbs action verbs for that so new new CS team's translation rules set get removed so forth these
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translation rules will affect the request URI from and to headers in the sip dialogue between the session border controllers and the service you will then create these roles and assign them to the Gateway the rules get assigned to the Gateway by utilizing either the new or the set C s online PSTN gateway command line and you can see the parameters that I have listed at the bottom here let us know what direction the translation rules
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come apply is it inbound or outbound and whether we're going to operate on the teams number or the PSTN number for that particular translation rule let's step through an example of number translation for an inbound call if we take a look at this screen here I have my session border controller which is configured with direct routing to Microsoft phone system to my service right and this session border controller is connected by a SIP trunk to a legacy
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PBX this could be our old you know Cisco or Avaya PBX that we're using I have a user Bob who is still utilizing his legacy PBX and then I have a user Alice who is configured with teams and direct routing and in this particular scenario Bob whose extension is a short four digit extension of two eight one four bob is gonna use his old handset to make a call to Alice now Alice isn't on the legacy PBX anymore but hey Bob doesn't know that all Bob knows is he wants to pick
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the phone up and dial two eight one four and Alice should answer when he calls Alice he knows to call Alice by her four digit extension which is four eight to five now in teams we will be using the e dot 164 notation to represent the users phone number so Alice isn't gonna have a phone number of four eight to five she's gonna have a phone number that's properly formatted as an e 164 number
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which would start with plus one likewise when Bob makes the call his caller ID should be reflected as he dot 164 as well so that Alice will have proper name resolution or what we call reverse number lookup and she'll actually see that it's Bob calling and not just extension 284 well in this instance the legacy PBX is using four digit dialing and teams is using EDA 164 so we're gonna need to translate the
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four digit numbers from Bob and Alice to eat out 164 now we need to pay attention to the direction that we're working with here because Bob is making the call to Alice so from the team's point of view this is an inbound call right the call is coming into the team's firemen to achieve this we need to create and apply a set of translation rules first I'm going to create two different translation rules here the one translation rule that I'm
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going to create it's going to simply add the plus one into the environment the second translation goal that I'm going to add is going to add the Seattle NPA and NX X values right that's the area code and so forth so you'll notice here for this second one when I see four digits I'm going to prepend it with plus one 206 which is theoria code 555 which is the exchange and that the dollar sign one that's the variable for the four
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digits right so these are my translation rules that I've created and then I'm going to take these rules and I'm going to apply them in a following fashion when I apply the translation rule for the inbound teams number right remember it this is inbound is to this direction and the number that I want to affect is Alice that means that I want to turn Alice's number from this four digit extension into a fully-formed
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be 164 number so I will use this inbound teams number translation rule and I will pick my rule of add a 160 for Seattle and PA and xxxx here's what happens on the request URI the invite from the PBX all it knows is hey I'm sending a call to 485 because that's what Bob dialed but on the team side when this invite reaches us we're gonna change that from four eight two five two plus one two oh six five five five four eight two five
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right we're gonna prepend this info because now this is a fully formed you want 64 number and when this makes it all the way through the team service we can see that that is Alice because that's Alice's fully formed phone number all right the same thing happens with the two requests because this is who the call is destined to write the two is the four digit number I use the translation role to make it fully formed is e.164 because I'm using that inbound team's number now the
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second piece of this was for caller ID right Bob's extension is two eight one for the PBX itself when it sends the call across this trunk here right it's actually sending it almost as a one 64 number right this is Bob's actual phone number nine four nine five five five two eight one four so it's close but it's not quite be one sixty four right it's only ten digits that I'm dealing with here so
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when I do this inbound because again remember it's inbound into teams right but now the number that I want to manipulate is the PSTN number right I want to manipulate Bob's number because Bob's number doesn't have a plus one on it so I'm gonna manipulate his inbound PSTN number I'm gonna add plus one and that's gonna translate his number into plus one nine four nine five five five two eight one four and now when Alice receives the call in the toast that she gets in her team's client
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she'll be able to have a reverse number lookup for that a 1/64 number match to the contact for Bob and now we know that it's Bob it's making the call now this is a way that we can achieve number translation through teams it's also possible to do these types of translations right here on the session border controller it's up to you where you choose to do this or maybe it isn't depending on if this is a hosted environment but the key to this is you only want to make these types of translations in one place either do them
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here on the session border controller or do them here inside of teams but generally speaking we don't want to do them in both places now that we have the session border controllers registered we have the voice routing configured the last step of the puzzle is to provision the user now the users can be provisioned to only use direct routing or they could be provisioned to use both direct routing and calling plans together let's start
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with direct routing only so the first column here again make sure that we have the correct licenses applied to our user where do we provision the phone number well when they're direct routing only we'll provision this in Azure Active Directory this will either be done online or you can do it through the on-premises environment with an azure ad sync if you like that might come into play if we were dealing with migrations from Sky for business server for example but the number is provisioned through a directory we can enable the user quite
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simply by using a set CS user command lit so here you stand this command let's set CS user my user name the - on-prem line URI is the phone number that we're going to assign to this user and this should be an e1 64 format - enterprise voice enabled true and - hosted voicemail true basically lights up the phone system direct routing and cloud voicemail components for them now this enables the user but there's no routing
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mechanism that's assigned to the user yet remember a couple slides ago we configured all of that voice routing but the last piece for that user now is we have to assign the voice routing policy to them and we do that through this grant CS online voice routing policy command lit that's my user and there's the policy that I would like to assign to that user now in this case in the first column remember from a routing behavior perspective only the routes that I've configured through this voice routing policy are evaluated so if
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there's no route that matches the user can't make a phone call and because there are only provision for direct routing that's their only way to make the call if the routes not there the call drops if we look at the second column this is where we configure this individual user with Bo direct routing and calling planets right so saint based licenses that we saw before but now we've also assigned a microsoft calling plan to the user and that would be a domestic or domestic plus international one of the main differences here in this scenario though
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is the number provisioning happens through Microsoft this is not a number that you assign through the direct routing environment this is a number that is acquired from the Microsoft service or a number that you port in to phone system in the Microsoft service so this is a number that gets assigned through the team's admin center or using the set CS online voice user command lit right not set CS user set CS online voice user or the team's admin center
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basically you are provisioning this user as a calling plan user you're just also giving them the ability to use direct routing for routing decisions but their calls inbound are anchored on the calling plan phone number now again you can assign the voice routing policy the same way you did with direct routing only here's the difference from an outbound routing perspective first thing that we do is check all the routes that are here in this online voice routing policy to see if we can route if we
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don't find any matches there well then we'll fall back to using calling plan and see if they have a domestic or international plan and if they do then we'll route the call out through calling plans this is very advantageous where perhaps you've given these are calling plans because it's more beneficial to them but maybe the user also has to still be able to do calls to an on-premises PBX this allows you to route those calls to the on-premises PBX through the direct routing environment but then route all other calls through
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the calling plan infrastructure if you want to look at the dial plans that we talked about in our planning session you can see on the left side here an example of PowerShell command that lets you see the base normalization rules that are in the service country dial plan this is an example of the service country dial plan for the US but you could substitute that with any particular dial plan like you see in the screenshot there you could then go into the team's admin center and create your own tenant level dial plan if you so
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desire like we discussed in our planning session you could either edit the global which again would apply to everyone in the tenant or you could click Add and build your own custom dial plan and then you could assign this particular dial plan to an individual user either through the UI here or through a grant CS tenant dial plan and then remember that you can use the get CS effective tenant dial plan to see what that merged dial plan would be for an end user now
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let's talk about configuring local media optimization from a configuration standpoint when we go to configure local media optimization we first need to ensure that we configure our trusted IP addresses properly this is to ensure that we understand when a user is inside the corporate network versus outside of the corporate network we also then want to be able to locate that user inside the corporate network which takes us to step two which is defining our network elements these are our regions our sites and our subjects from there we'll define
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the virtual topology which is basically configuring the proxy session border controllers as well as any downstream session border controllers from there we then need to configure the SPC's themselves for local media optimization that is vendor specific guidance so we need to make sure that we get the correct configuration information from our SBC vendor of choice and then finally we would configure the voice routing pieces which is exactly the same as we had seen before which is the ability to just assign a online voice
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route to a particular session border controller as the target now as I go through these steps I'm gonna use a configuration example as my guide and this example you might recognize from our planning session this is our a pack example and as you can see in this configuration example we have three sites we have our Singapore site which is the site that actually has Internet connectivity as well as the
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public IP address for the session border controller and then we have two additional branch office sites Vietnam and Indonesia as you see here these sites also have session border controllers in them but these session border controllers don't actually have public IP addresses assigned to them now these particular sites the branch offices of Vietnam and Indonesia they do have local internet breakout right so from an end-user perspective I can browse the web but there was no capability for us to assign a public IP
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directly to these session border controllers from a when connectivity perspective the Vietnam and Singapore sites and the Indonesia to Singapore site that's good connectivity right but between Indonesia and Vietnam itself that's not so great connectivity so we want to keep that in mind as we start looking through the configuration steps over the next set of slides so our first step is to configure the trusted IP
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addresses the trusted IP addresses are the IP addresses that the users would be using when they browse the web and actually hit the office of sixty five service so each of the respective sites that we looked at Vietnam Indonesia and Singapore they all had local internet connections right so when our users would connect to the service with their teams clients these are effectively the NAT and IP addresses public IPS that we would see for these users so I would add them either using PowerShell as you see here UCS tenant trust if IP address or I
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could do this through the teams admin Center under network topology flip over to the trusted IP is tab and click Add and away you go important thing to note here when I'm adding these addresses I using a mask bits of 32 that indicates that I'm matching this exact IP address now of course that's not a public IP but you get the point this would actually be a public IP address that your users would be using when they egress out that internet connection our next step is to configure
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the Associated network elements and again these are the same elements that we would use if we were doing location-based routing or potentially dynamic emergency calling so if you've already configured those you might be able to reuse some of these components if you haven't then we simply need to define regions sites and subnets so you can see here I built out my region as APEC I built out my sites as for the example right I have my Vietnam site I have my Indonesia site and I had my Singapore site and you notice that all three of them are tied to the same region tied to the APEC
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region and then I built out my appropriate Network subnets CS tenant Network subnet these are my subnet IDs masks and the network site ID that they are associated with now a key piece of the puzzle here the network configuration service has roughly to our cache window so newly created configuration settings like trusted IPS or things of that nature may not be directly available until roughly two hours after creation so just keep that in mind as a buffer if you're
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doing some testing and you go and create one of these things and you don't see an immediate response it might be because the NCS cache hasn't flipped yet so you haven't been able to see that you change the email our next step is to define the virtual topology itself this is the configuration of the proxy SBC's and the downstream SBC's now remember in the example that we were looking at there's a total of three SPC's the one that's in Singapore that's my proxy SBC because he's the SBC that actually has a public IP address
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assigned to him my other SBC is our downstream SPC's in Vietnam and Indonesia respectively now the proxy SBC it's defined pretty much the same as we had seen defining session border controllers before we use the new CS online PSTN gateway command line we give it the fqdn the port that it's going to use for signaling the max concurrent sessions great thing to do the gateway site ID right that's the
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matching site that we built and this is case sensitive so make sure that you match the case that you did here with the case that you did when you created the site you know on the previous steps the media bypass flag needs to be set to true right that's in our planning session we kind of talked about the on/off switch for media bypass that's the own off switch right we do bypass is set to true - bypass mode this is always in this particular example right remember we
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talked about in our planning session this could be either always or it could be only for local users so we've set this guy to always because he has good connectivity and of course don't forget to enable it - enabled true now this SVC might already be defined in the topology maybe you just didn't turn on media bypass that's cool you could also use the set CS PSTN gateway command line and add the you know gateway site ID media bypass and bypass mode options as appropriate our downstream SPC's this is the ability
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for us to effectively build that virtual topology let the service know that these SBC's exist even though they don't have a public IP so do see us online PSTN gateway similar command write the identity that's the fqdn if you will of that particular device the signaling port that it's using again that's how you figured what you chose on the SPC this is the key part - proxy SBC and then the fqdn of the proxy SPC this lets us know that this is
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a virtual definition of a gateway I'm not gonna try to talk to him directly but if I need to talk to him I talked to him through this proxy SBC gateway site ID again case sensitive media bypass has to be true bypass mode well in this case I've chosen only for local users because I don't have great connectivity between Vietnam and Indonesia so if I have a user that's in Indonesia and they want to send traffic out through the Vietnam gateway I don't want them to talk to me directly so you have to be in the same
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site as this session border controller to be able to talk to it directly and I do the same thing here for the Indonesia session border controller right again it's important to note Gateway site ID is case sensitive so as you see here Vietnam with a capital V and Vietnam with a lowercase V would be treated as different sites so be aware of that the domain that you use for the downstream SPC's right here I have Kanto SOCOM that actually can be a domain that maybe isn't even configured in the tenant
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right maybe it's you know the VN SBC contoso dot local for whatever reason that's fine you can use that as long as you use the dash proxy SVC parameter if you forget this and you try to put a name in here that doesn't match the domain list that we have in the tenant you're gonna get an error all right and again remember with the only for local users option the bypass mode right local media candidates are only going to be offered to the team's client when that team's client is in the same location as
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the SVC right the same defined site as the SVC in all other cases media is going to flow via the internal or the external IP address of the proxy SBC and again that's going to be determined on whether the user is internal right if the user is internal it'll flow by the internal IP of the proxy SBC if they're external then it's going to throw it flow by the external IP address of the proxy SBC from there you're going to make sure that you follow the vendors
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documentation to properly configure the session border controller for local media optimization and then you'll go ahead and configure voice routing as outlined in our voice routing sections of our planning and our configuration modules right these downstream SBC's will show up as gateways right you built your routes you'll point it to these gateways and away you go you now let's talk about the tool sets we have for managing director outing when we look at the direct routing environment as we've talked about before
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we have our session border controllers we have the PSTN Network perhaps we even have a third party PBX in the mix here how do we look at the environment and manage what's going on here well first and foremost if I'm looking at the connectivity between teams and the session border controller itself I can use tool sets such as the call quality dashboard or the call analytics now the difference between the two call quality dashboard which you can find at the URL
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that you see here call quality dashboard is a near real-time dashboard that allows us to see the calls that are being made by a user containing all the end-user identifiable information this is data that's aggregated as well so that we can start to see trend analysis of calls that have been made by the director adding infrastructure among other things right this does contain and user identifiable information such as the wireless access point BSS idea they were connected to their call details or
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user IDs or fully P's max and so forth there's lots of great information that you can drill down into utilizing the call quality dashboard plus it gives you that aggregated view of calls over time the call analytics tool on the other hand is a tool that we can use to dig deeper into an individual call write call reliability and availability for an individual call so maybe a user indicated that I made a call yesterday at 2:00 in the afternoon that particular
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call had some poor quality I could use call analytics to drill deep into that one specific call additionally because the session border controller could be talking to other environments we have logs that we can pull from the session border controller itself to help us debug or troubleshoot any particular issues we might be having and if the session border controller happens to be talking to say a third party PBX there might be logs that you need to collect
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from the third party PBX as well lastly we have our PS tn providers which you may need to dig a little deeper into with your carrier of choice to get some additional visibility into the PSTN provider side of the equation in the service we do provide a health dashboard to look at the overall health of our connected session border controllers often this is the first place to start because if the session border controller isn't successfully connected to the service nothing else is
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going to matter because we're not going to be able to make or receive calls through that session border controller so as you can see in the screenshot if I navigate to the voice blade and go under direct routing I will be able to see all of the session border controllers that I have as well as which session border controllers might have issues right and each of these are hyperlinks that we could go through to see what these particular issues might be whether it's with sip options or they're it's a failure with TLS connectivity that's our signaling channel right whether our network effectiveness rate or NER rate
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is below a certain threshold right if we've met our current capacity for calls we can very easily at a glance see what's going on with our session border controllers I can then click on an individual session border controller and go deep and see with that particular SBC when did it last connect in with TLS am I getting and receiving options properly what's my NER what's my network effectiveness ratio the average collaboration and so forth so lots of good telemetry and detailed information
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that we can get even as we drill down into the respective SBC's if you look at the AKMs link in the upper right you can see our published documentation on the health dash this is a table that shows some of the more common configuration issues that we see with a session border controller not going to take you through every single one of these but some of them are common ones we see are for example the second one fail to verify peer certificate the session border controller infrastructure
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utilizes TLS mutual TLS right to ensure that if the person on the other end of the device on the other end is who they say they are when your session border controller is talking to the office 365 service our service uses certificates that are actually generated by a certificate authority that uses the Baltimore certificate if you forget to import that certificate into your SBC you may not trust the certs that are coming from the service and that's why you'll get this failed to verify peer
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certificate all of the vendor documentation for the various s pcs have a step that talk about how to import this Baltimore root certificate for example you may also see errors these are errors you might see in SIP logs on the session border controller like 404 not found for incoming calls that usually means that we can't find the phone number so either we didn't configure the user's phone number correctly in the service so if we were configuring them with the on-prem line URI parameter perhaps we had a typo in
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that or we're not translating it properly maybe we forgot to include the plus sign or more fully form it as you dot one sixty four and the number just simply doesn't match now inside of the call quality dashboard we have also exposed a set of power bi reports these power bi reports include a set of summary reports the various helpdesk reports location enhanced reports mobile device reports and for
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the purposes of this training PSTN direct routing reports that let you drill into more and more detail of all the calls that were made through the PSTN direct routing infrastructure and then there's also a user feedback report these reports are available through the call quality dashboard portal itself which is at CTD teams Microsoft comm but they can also be downloaded to be used with the power bi desktop application so if you go to the AKMs link that is on
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the slide here that will take you to a page where you can find a hyperlink to download these PBX files that you can use with power bi desktop very powerful way to gain access to the power bi reports without having to navigate to the online version of CD let's summarize what we've talked about in our direct routing sessions our key learnings direct routing allows us to connect our existing phone infrastructure to the Microsoft teams environment to successfully configured
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direct routing requires a certified session border controller and we've covered the key pieces of configuration for direct routing but we have lots of documentation that go into great detail on all of the additional configuration that you may want to do for the direct routing environment in this lesson we completed the third and fourth bullet points on the slide which were configuring and managing director outing now that you've completed this lesson
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you should move on to our next lesson which is migrating to direct routing

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