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Cisco VoIP CCNP Voice CIPT2 v8.0 642-457

Cisco Voice over IP

Course Duration: 09:01:36
CUCM Multisite: Understanding the Big Picture Design
You can think of this Nugget as the "course overview." Jeremy walks through the major issues you can encounter when attempting to deploy CUCM in a multisite environment. Specific topics include CUCM connections, Quality of Service (QoS), bandwidth control, dial plan considerations, availability (strategies such as Survivable Remote Site Telephony (SRST)), and device/extension mobility.
Multisite Connections: Connection Options
When connecting outside the CUCM Cluster you have many options: H.323 Gateways, MGCP Gateways, SIP Trunks, and Intercluster Trunks. In this Nugget, Jeremy walks through the primary usage spots and advantages of each connection option.
Multisite Connections: Implementing H.323 and MGCP Gateway Connections
It's time to configure the two primary gateway protocols in CUCM: H.323 and MGCP! In this Nugget, you'll walk through the IOS and CUCM configuration of analog voice ports using both H.323 and MGCP.
Multisite Connections: Implementing SIP and Intercluster Trunk Connections
While H.323 and MGCP are primarily used for Cisco Voice Gateway connections, SIP trunking is used for just about everything else. In this Nugget, Jeremy walks through the configuration of SIP Trunks (and discusses connections to ITSPs for lab environments) and Intercluster trunks (which connect multiple CUCM clusters).
Multisite QoS: Addressing QoS Concerns
Quality of Service (QoS) is a MASSIVE concern in multisite CUCM environments, however it's officially a topic of the CVOICE series (also available here at CBTNuggets). Nonetheless, Jeremy goes "all in" to cram as much useful QoS information as he could in this ultra-informative Nugget!
Multisite Dial Plan: What is a Partition and CSS
Partitions and Calling Search Spaces (CSSs) are HUGE, similar to Access Control Lists (ACLs) in Cisco Routers. Because this concept is used so heavily in an enterprise CUCM dial plan, Jeremy created a fly-by review (from CIPT1) of the concept here.
Multisite Dial Plan: Dial Plan Scalability and Site Codes
In this thrilling Multisite Dial Plan episode, Jeremy discusses the all-to-common issue of overlapping phone numbers and how you should handle them in both a distributed and centralized CUCM design. We're not too sure why, but after this Nugget, we were all very excited to play with phone numbers.
Multisite Dial Plan: PSTN Access - Handling Incoming and Outgoing Calls
Even though the CIPT2 series is focused on multisite scenarios, understanding the PSTN access for a single site is absolutely critical before you can build multiple sites upon it. In this Nugget, Jeremy discusses handling incoming and outgoing calls (including ANI, DNIS, and ISDN TON) on digital (T1/E1) and analog (FXO) circuits.
Multisite Dial Plan: PSTN Access for Multiple Locations
Building upon the foundation created in the "PSTN Access - Handling Incoming and Outgoing Calls" Nugget, Jeremy discusses the design of a dial-plan for multiple office locations in a Centralized CUCM design covering both the traditional design and using Local Route Groups.
Multisite Dial Plan: Implementing Intersite PSTN Backup
One advantage of using VoIP connections between your offices is the ability to route calls over the WAN connection, avoiding toll charges; but what happens when that WAN connection goes down? That is precisely the focus of this Nugget! In here, you'll revisit the CUCM route plan design to discuss the digit manipulation necessary to move calls to the PSTN in an outage situation.
Multisite Dial Plan: Tail-End Hop-Off (TEHO)
Tail-End Hop-Off (TEHO) is one of the fantastic, toll-charge-avoiding methods you can use in a multisite organization. Join Jeremy as he looks at the implementation of TEHO in a traditional and Local Route Group-based design.
Multisite Dial Plan: CUCM Globalization
Despite countless attempts, there really is no good explanation of CUCM Globalization on the Internet . . . until now! Join Jeremy in this ultra-informative Nugget as he discusses the E.164 standard, the + sign in dial strings, and the goals and techniques of using globalization and localization in CUCM.
Multisite Dial Plan: Understanding MGCP Fallback and SRST
Help your IP Phones defy death! In this Nugget, you will see the various failover options for your VoIP Network (MGCP Fallback, SRST, and CME, SRST Mode), the method and timers that IP Phones use to failover (and defy death), and SRST router sizing.
Multisite Dial Plan: SRST, CUCME, and MGCP Fallback Configuration
The first step in helping your IP Phones defy death is to configure the CUCM server appropriately. In this Nugget, Jeremy discusses the SRST Reference, Device Pool, and Call Forward Unregistered (CFUR) configuration.
Multisite Dial Plan: SRST, CUCME, and MGCP Fallback Configuration, Part 2
Once you've configured your CUCM server for SRST, it's time to focus on the router itself. This Nugget dives into IOS-based call-manager-fallback configuration. Topics include base configuration, MGCP fallback, dial-peers, translation-rules, and translation-profiles.
Multisite Dial Plan: SRST, CUCME, and MGCP Fallback Configuration, Part 3
This final Nugget in the three part SRST configuration trilogy focuses on using a CUCME (CallManager Express, often abbreviated CME) configuration in an SRST setting. You'll not only learn how to configure CUCME as a standalone phone system, but will also learn how to use auto-provisioning (and SNAP) to maintain phones in a centralized call processing model.
Multisite Bandwidth: Strategies for Addressing Bandwidth Concerns
Anytime you talk about Multisite deployments, bandwidth management must enter the discussion. This Nugget discusses "big picture" bandwidth strategies and walks through QoS and CAC definitions, Region (CODEC) configuration, local conference bridge (CFB), Music on Hold (MOH), and Transcoder (XCODE) configuration, along with the configuration of DSPFarm resources on a router. Now THAT'S a LOT of stuff!
Multisite Bandwidth: Call Admission Control - Dial-Peer Limitations and CUCM Locations
While QoS protects voice from data, what if you have to protect voice from voice? The journey of bandwidth management has led us to Call Admission Control (CAC). This Nugget focuses specifically on dial-peer based restrictions and CUCM Locations (both basic and RSVP-Enabled Locations).
Multisite Bandwidth: Automated Alternate Routing
Up to this point, we've shown you how to DENY VoIP calls going across the WAN using various CAC methods, but we have yet to show you what to do with those denied calls...until now! In this Nugget, Jeremy breaks down Automated Alternate Routing (AAR), which allows calls to re-route over the PSTN when the WAN bandwidth is constrained.
Multisite Bandwidth: Gatekeeper and SIP Preconditions
We're winding down the Multisite Bandwidth portion of CIPT2 with a focus on distributed CAC. This allows you to implement CAC between completely different voice systems (different vendors, administrative authorities, etc...). In this Nugget, Jeremy explains the key concepts and configuration of H.323 Gatekeepers and SIP Preconditions.
SAF and CCD: Understanding How SAF and CCD Works
If you're one for "revolutionary technology," you'll absolutely love this Nugget! The Service Advertising Framework (SAF) and Call Control Discovery (CCD) protocols represent an evolution in designing voice network by allowing the Call Agents (and SAF Forwarders) to do the dial plan work for you. Tune into this Nugget to learn more!
SAF and CCD: Configuring SAF and CCD
SAF and CCD configuration...there's a LOT to it! In this Nugget, Jeremy breaks down the configuration of SAF and CCD into three sub-sections: the Configuring SAF Forwarder, Configuring CUCM as an External SAF Client, and Configuring a SAF Internal Client.
Device Mobility: Device Mobility Concepts and Configuration
It's a problem we never experienced in the PBX days: a phone that MOVES from site to site. Because of the havoc this can cause with Device Pool configurations (regions, locations, etc...), Cisco created Device Mobility! In this Nugget, Jeremy walks through the complete concept and configuration of Device Mobility for centralized CUCM deployments.
Device Mobility: Extension Mobility Concepts and Configuration
The base Device Mobility feature is very handy if you'd like to move a phone from one site to another...but what if you'd rather not move the phone? That's where the Extension Mobility feature comes in! In this Nugget, you'll learn how to allow your users to log into their phones and assume whatever phone profile you've associated.

No Bookmarks

The CIPT2 video training series with Jeremy Cioara covers the advanced features of Cisco's Unified Communication Manager (CUCM), focusing on the challenges associated with a multisite VoIP deployment.

Recommended skills
  • Knowledge from CIPT1 8.0 highly recommended
  • Knowledge from CVOICE 8.0 recommended
  • Previous hands-on experience with Cisco CUCM

Recommended equipment:
  • Cisco Voice Gateway (Cisco 2621XM or Cisco 2801 are great, low-cost lab routers)
  • CUCM lab server (Virtualize in VMware ESX or VMware Workstation)

Related certifications:
  • CCNP Voice

Related job functions:
  • VoIP Technician
  • CUCM Administration

This CIPT2 video training course forges the brave world of a multi-site VoIP deployment with CUCM. Because these worldwide deployments can be flat-out overwhelming, the course walks through the design from the ground up. You will learn connecting multiple sites using voice gateways and intercluster trunks, designing and implementing a globalized dial-plan, implementing call processing redundancy, device mobility, and more!

CUCM Multisite: Understanding the Big Picture Design

00:00:00 -
00:00:00 - Understanding the big picture design.
00:00:02 - Or I think I'd rather call this CIPT2 Series Overview.
00:00:06 - Because that's what this really is.
00:00:08 - We're going to be looking at multi-site deployments and the
00:00:11 - challenges associated with them.
00:00:13 - Now, it makes it a very awkward Nugget to record,
00:00:16 - because I'm going to be going through each one of these
00:00:17 - challenges, like what kind of connections do you bring up,
00:00:19 - quality of service, bandwidth concerns.
00:00:22 - All those kind of things.
00:00:23 - I'm just going to be listing these challenges and not give
00:00:25 - you any solutions, because that's what the rest of this
00:00:27 - series is going to be about.
00:00:29 - So I'm going to say, there's a problem, so we've got to deal
00:00:31 - with that, and then move right on from there.
00:00:33 - So the first thing that we have to think about is, what
00:00:36 - about our connections?
00:00:37 - Let's look at our topology.
00:00:38 - We've got Phoenix over here, which looks like it's got its
00:00:41 - own little call manager-- we'll say
00:00:42 - its publisher server.
00:00:43 - Its own cluster.
00:00:44 - Same thing with Australia.
00:00:45 - It's not a cluster split over the WAN.
00:00:48 - This is another publisher server.
00:00:49 - Completely separate database all together.
00:00:52 - San Diego is using that centralized deployment model,
00:00:55 - so these phones get their dial tone from this call manager
00:00:58 - down here in Phoenix.
00:01:00 - Now if the WAN link goes down, they have some kind of fail
00:01:03 - over that's going to be a little bit of what we talk
00:01:04 - about in this series.
00:01:05 - Actually, a big part of what we talk about.
00:01:07 - This router's going to be needing to run something like
00:01:09 - SRST, survivable remote site telephony, so that way they
00:01:13 - still get dial tone, even if the WAN link is offline, and
00:01:15 - maybe can even make PSTN calls.
00:01:18 - I say maybe because PSTN is now becoming almost optional
00:01:23 - in each one of these locations.
00:01:24 - Kind of, but not really, but sort of is.
00:01:26 - But we'll talk about that.
00:01:28 - So connections means, how do you get all these things
00:01:30 - connected together?
00:01:31 - For example, you want to take Phoenix, which maybe has one
00:01:35 - xxx extension.
00:01:35 - So 1,000 and 1,001.
00:01:37 - All that.
00:01:37 - And then Australia, which is, we'll say 21xx extensions.
00:01:41 - And you want to make sure that these guys can call each other
00:01:43 - without paying the toll charges by going
00:01:45 - over the WAN link.
00:01:46 - How do you do that?
00:01:47 - Well, I mean, there's different options.
00:01:49 - You could say, well, I want to use a SIP trunk to do that.
00:01:52 - Or you might say, well, actually, I'd rather link
00:01:54 - these call managers directly using an intercluster trunk.
00:01:57 - But then someone comes along and says, well, did you know
00:01:59 - the gateways could do that?
00:02:00 - You could actually point straight to that gateway.
00:02:02 - And to do that, you could use H.323, or you could use MGCP.
00:02:06 - And that's always preferred, because Cisco
00:02:07 - developed that one.
00:02:08 - But oh, you can also use SIP as well.
00:02:10 - And then the gateways can point to each other, and they
00:02:12 - kind of manage all your outside dialing plan.
00:02:14 - And then you can throw in a gatekeeper,
00:02:15 - which as you grow--
00:02:17 - I mean, see what I mean?
00:02:18 - It's like, huh?
00:02:20 - Well, OK.
00:02:21 - So you're saying there's a lot of ways to do this?
00:02:23 - Yes.
00:02:24 - Which one's best?
00:02:25 - Well, it depends.
00:02:26 - So that's going to be what we talk about as we go through
00:02:28 - this series.
00:02:29 - Then we drop down into quality of service, which is I know a
00:02:31 - topic that you've all heard about.
00:02:33 - And I will say out of all the topics, we will touch on this
00:02:36 - one lightest in here, because it's actually part of the
00:02:39 - CVOICE series, one of the prerequisite series at CBT
00:02:42 - Nuggets before you would have gotten here.
00:02:45 - However, it's still worth mentioning, because you want
00:02:47 - to make sure that when voice goes across the WAN, and
00:02:50 - there's data traffic, voice always wins.
00:02:52 - That's the whole goal of quality of service, is to
00:02:54 - prioritize some traffic over others.
00:02:57 - But actually, as we bleed between quality of service and
00:03:00 - bandwidth, we'll start talking about another challenge, which
00:03:03 - is, let's say you've got Fred in Australia--
00:03:06 - are there people named Fred in Australia?
00:03:08 - I don't know.
00:03:09 - But Fred calls over here to Sally in Phoenix.
00:03:12 - And then, Beth does the same thing, and Ron
00:03:15 - does the same thing.
00:03:16 - And you start getting all these calls.
00:03:17 - And now it's no longer a voice versus data match, which is an
00:03:22 - easy victor when you set up quality of services.
00:03:24 - It's like data, dum, hits the mat.
00:03:26 - Voice is the one that ends up winning.
00:03:28 - But what do you do when you have voice versus voice?
00:03:33 - The biggest fighter wins, right?
00:03:34 - Whoever talks loudest wins.
00:03:36 - Actually, no.
00:03:37 - As we bleed into this bandwidth control, we'll
00:03:39 - introduce a topic called call admission control, which says,
00:03:42 - OK, I can only have so many voice calls going across that
00:03:46 - WAN before it's just flat out of bandwidth.
00:03:48 - Quality of service is so fun because with voice, you're
00:03:52 - like, oh man, it gets the priority, but it's actually
00:03:54 - called strict priority, which means if voice ever tries to
00:03:58 - go above its priority bandwidth limit that you set,
00:04:02 - it's like quality of service is like, no go.
00:04:03 - No.
00:04:03 - I'm not going to try.
00:04:05 - I don't care if I have 1,000 megabits per second sitting
00:04:07 - there idle.
00:04:08 - You're dead voice.
00:04:09 - You're not getting any further, which is kind of fun.
00:04:12 - And that's where call admission
00:04:13 - control becomes essential.
00:04:14 - So we're going to look at bandwidth control as a big
00:04:17 - picture for that.
00:04:18 - Now, when you're talking about bandwidth control, let me get
00:04:20 - rid of all this gibberish here, bandwidth control says,
00:04:23 - I want to effectively use my resources.
00:04:26 - For example, let's say that we have Sue over here in San
00:04:29 - Diego talking to Beth on the phone.
00:04:34 - And Sue says, oh, hang on one second Beth, and presses the
00:04:37 - Hold button.
00:04:38 - Well, I don't know about you, but I don't see any music on
00:04:42 - hold at juke boxes sitting in here in San Diego where it's
00:04:45 - like, oh, this mystery box starts sending music notes
00:04:48 - down to Beth.
00:04:49 - No.
00:04:49 - What happens in a centralized deployment is this guy is the
00:04:52 - music on hold server.
00:04:53 - We talked about that in the CIPT1 Series.
00:04:55 - In the Media Resources, we said, OK, well, one of the
00:04:58 - call managers will be designated as a
00:05:00 - music on hold server.
00:05:01 - And it's going to be streaming.
00:05:02 - So as soon as Sue presses the Hold button for Beth, you're
00:05:06 - streaming music across the WAN link from
00:05:08 - Phoenix over to San Diego.
00:05:10 - There's got to be a better way.
00:05:11 - Same thing, what if we do a conference call?
00:05:13 - We have Sue and Beth--
00:05:15 - and there's more phones here.
00:05:16 - I just didn't draw them.
00:05:18 - And Joe, and Bob, all these people jump on
00:05:21 - a conference call.
00:05:21 - Again, I don't see any conference bridges other than
00:05:25 - Phoenix over here.
00:05:26 - Conference bridge, media resource.
00:05:27 - So all these guys start hopping down here to Phoenix.
00:05:30 - You're having a conference call over the WAN link, when
00:05:33 - all the people on the conference call are
00:05:34 - sitting in San Diego.
00:05:35 - Again, not a good use of bandwidth.
00:05:37 - So we'll look at how we can circumvent
00:05:39 - those kind of things.
00:05:40 - Now, dial plan.
00:05:41 - Oh, my word.
00:05:42 - If there is a concept that is probably more complex than any
00:05:45 - of them, this would be it.
00:05:46 - When we went from the PBX world to the voice over IP
00:05:50 - world, it just exponentially became complex.
00:05:54 - Because in the PBX world, you essentially said, I have one
00:05:57 - exit point.
00:05:58 - It is the PSTN.
00:05:59 - Maybe you had an analog backup or something to the T1, which
00:06:03 - added a little complexity.
00:06:04 - But now, I mean here's our goal.
00:06:05 - Let's say we've got, going back here to Phoenix, 1xxx,
00:06:08 - and we've got 21xx.
00:06:10 - And both of these locations, Australia and Phoenix, have a
00:06:13 - good T1 line to the PSTN.
00:06:16 - That's their PSTN access.
00:06:18 - But I don't want to pay the long distance to call between
00:06:22 - my offices over that.
00:06:23 - So when I dial 2102, which is this phone right here, it's
00:06:28 - going to go over the WAN, which I want.
00:06:29 - But what happens if the WAN is down?
00:06:32 - Then do I just lose Australia calling?
00:06:34 - Do I have to now send out a notice to the company?
00:06:37 - Email, hey FYI, WAN link down.
00:06:40 - Everybody dial international numbers to reach Australia
00:06:44 - now, because the WAN link--
00:06:46 - oh, no, it just came up.
00:06:47 - Email.
00:06:48 - No.
00:06:48 - No, what we want is we want this set up to where when I
00:06:51 - dial 2102, and call manager's like, oh, I know
00:06:54 - the WAN link is down.
00:06:56 - I'm going to automatically transform that number into the
00:06:58 - valid PSTN number.
00:06:59 - You know, stick on 011, and then country code.
00:07:02 - I mean, I'm talking United States dialing plan here.
00:07:05 - But you transform the number to go across the PSTN.
00:07:08 - And there's no way I'm going to show 1010 as the caller ID,
00:07:13 - because that doesn't work over the PSTN.
00:07:15 - So I also want to transform that caller ID to work across
00:07:17 - the PSTN as well.
00:07:19 - You see what I mean?
00:07:20 - Now, OK.
00:07:21 - You're like, OK, I can hang with that.
00:07:24 - I'm following that.
00:07:25 - You go, well, I want to introduce also a topic called
00:07:27 - TEHO, tail and hop off, which says, you know what?
00:07:30 - I bet it's a lot cheaper for Australia to
00:07:33 - call Australia numbers.
00:07:34 - And it's a lot cheaper for Phoenix to
00:07:36 - call Phoenix numbers.
00:07:37 - Right?
00:07:37 - So why don't I set this up to where, for instance, when
00:07:40 - someone in Australia dials--
00:07:41 - I don't know what they dial for an
00:07:43 - international number in Australia.
00:07:45 - I'll just say 1-602, because it's what we dial here in the
00:07:47 - United States.
00:07:48 - 555-1111, which is a number out here in Phoenix, because
00:07:53 - we have the 602 area code.
00:07:55 - Well, why not instead of just paying that toll charge to
00:07:58 - call that, why not route that across the WAN to Phoenix, and
00:08:02 - then Phoenix can send that out to the PSTN, and I don't pay
00:08:05 - any toll charges at all.
00:08:06 - And the bigger and bigger my company grows, the more
00:08:08 - locations I get all around the globe, the more free stuff.
00:08:12 - It's kind of like, woohoo!
00:08:14 - As we grow, we got all these expenses.
00:08:15 - But you know what?
00:08:15 - At least we get free long distance to whatever
00:08:18 - area that that is.
00:08:19 - That's called tail end hop off.
00:08:20 - And then you say, well, but I only want certain people to be
00:08:24 - able to dial that.
00:08:24 - I don't want everybody to be able to dial those numbers.
00:08:26 - Maybe partitions and--
00:08:28 - [WHOOSH].
00:08:30 - That was my head.
00:08:31 - It just exploded.
00:08:32 - Seriously, that's the dial plan
00:08:34 - complexity that we get into.
00:08:35 - But I will say once you get through the dial plan Nuggets
00:08:39 - that are going to be part of this series, you're going to
00:08:41 - have the knowledge necessary to create that
00:08:43 - crazy complex outline.
00:08:45 - And this is a whiteboard on my whiteboard.
00:08:47 - Because a whiteboard is required for this.
00:08:50 - It's seriously one you will spend hours on.
00:08:53 - And you'll sit there and go, OK, well, if this person dials
00:08:55 - here, then this, and then this.
00:08:56 - But the fun thing is, after this series, you're going to
00:08:58 - have the knowledge to do it, to where it becomes like
00:09:00 - putting together a fun jigsaw puzzle rather than
00:09:03 - Googling for hours.
00:09:04 - And Bob just advised me wrong, and now I have
00:09:08 - to do it all over.
00:09:08 - I mean, you don't want to do all those things.
00:09:10 - So the dial plan is a big part of what we're
00:09:13 - going to talk about.
00:09:13 - Availability, we already talked about that.
00:09:15 - That's keeping the site running.
00:09:16 - Should it go down, you have SRST, survivable remote site
00:09:19 - telephony, to do that.
00:09:20 - And you can even run it in Call Manager Express Mode.
00:09:24 - And what that is giving SRST more features, more ability,
00:09:29 - than the base SRST feature set.
00:09:34 - And then, finally, mobility.
00:09:35 - Mobility is what happens when people move.
00:09:37 - Now, that's one of the beauties of this voiceover IP
00:09:40 - phone system, is you just unplug the phone and move.
00:09:43 - But that also creates a little chaos when somebody grabs
00:09:47 - their phone and flies to Australia, and then dials an
00:09:49 - emergency number, and then all the calls end up here in
00:09:53 - Phoenix, and all that kind of stuff.
00:09:54 - So we're going to talk about device mobility.
00:09:55 - We'll talk also about extension mobility.
00:09:57 - That's kind of like roaming profiles, to where somebody
00:10:00 - can log into a phone here in Phoenix, and then they don't
00:10:02 - have to bring their phone with them.
00:10:03 - They just walk, or fly, to Australia, and then log into a
00:10:07 - phone here.
00:10:07 - And their whole profile, their whole phone, kind of follows
00:10:10 - them along.
00:10:12 - These are the considerations that we need to have in mind
00:10:14 - as we move forward with a multi-site deployment.
00:10:16 - All that to come in this series.
00:10:18 - I hope this has been informative for you.
00:10:20 - And I'd like to thank you for viewing.
00:10:21 -

Multisite Connections: Connection Options

Multisite Connections: Implementing H.323 and MGCP Gateway Connections

Multisite Connections: Implementing SIP and Intercluster Trunk Connections

Multisite QoS: Addressing QoS Concerns

Multisite Dial Plan: What is a Partition and CSS

Multisite Dial Plan: Dial Plan Scalability and Site Codes

Multisite Dial Plan: PSTN Access - Handling Incoming and Outgoing Calls

Multisite Dial Plan: PSTN Access for Multiple Locations

Multisite Dial Plan: Implementing Intersite PSTN Backup

Multisite Dial Plan: Tail-End Hop-Off (TEHO)

Multisite Dial Plan: CUCM Globalization

Multisite Dial Plan: Understanding MGCP Fallback and SRST

Multisite Dial Plan: SRST, CUCME, and MGCP Fallback Configuration

Multisite Dial Plan: SRST, CUCME, and MGCP Fallback Configuration, Part 2

Multisite Dial Plan: SRST, CUCME, and MGCP Fallback Configuration, Part 3

Multisite Bandwidth: Strategies for Addressing Bandwidth Concerns

Multisite Bandwidth: Call Admission Control - Dial-Peer Limitations and CUCM Locations

Multisite Bandwidth: Automated Alternate Routing

Multisite Bandwidth: Gatekeeper and SIP Preconditions

SAF and CCD: Understanding How SAF and CCD Works

SAF and CCD: Configuring SAF and CCD

Device Mobility: Device Mobility Concepts and Configuration

Device Mobility: Extension Mobility Concepts and Configuration

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Jeremy Cioara

Jeremy Cioara

CBT Nuggets Trainer

Cisco CCNA, CCDA, CCNA Security, CCNA Voice, CCNP, CCSP, CCVP, CCDP, CCIE R&S; Amazon Web Services CSA; Microsoft MCP, MCSE, Novell CNA, CNE; CompTIA A+, Network+, iNet+

Area of Expertise:
Cisco network administration and development. Author or coauthor of numerous books, including: CCNA Voice 640-461 Official Cert Guide; CCNA Voice Official Exam Certification Guide (640-460 IIUC); CCENT Exam Prep (Exam 640-822); CCNA Exam Cram (Exam 640-802) 3rd Edition; and CCNA Voice 640-461 Official Cert Guide.

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