Archive for July, 2009

Understanding RIP Version 1

In our last post we discussed the basics of Router Information Protocol (RIP) such as the metric and the timers.  We even saw a few examples of debug and show commands to understand some of the processes that occur behind the curtain with RIP.  However, for the CCNA exam you must also have good grasp on understanding RIP version 1.

First, RIP version 1 is a classful routing protocol.   Primarily, classful routing protocols do not advertise the subnet mask with network addresses inside of routing updates.  As shown in example 1, the debug ip rip command is running on two routers, R1 and R2.  We can see here that R1 is sending a version 1 update without the subnet mask for any network entry. We also see this being received on R2.  Also, to be a classful routing protocol you will always summarize to the classful boundary between routers that have a different assortment of networks.  Example 1 displays that the update has class A, class B, and class C networks and not its subnets. Continue reading ‘Understanding RIP Version 1′

Basics of Understanding RIP

RIP is a protocol that is used for routing IP networks. It was designed in the early 1980’s for communication between gateways (computers with two NIC’s). It is the oldest routing protocol used by the network industry and is considered by many to be inefficient or border-line obsolete.  However for CCNA students it important to understand RIP, as well as how to configure and troubleshoot it. This post covers RIP’s baseline features.

First, the metric for RIP is hop count, where a hop is a router or gateway. When a router or gateway receives a packet, processing required for the delivery of the packet would insert latency (Processing Delay). The delay is cumulative, so the best path will always be the chosen based on fewest number of hops. However due to this delay, RIP has a maximum hop count of 15, so 16 hops is considered inaccessible. Continue reading ‘Basics of Understanding RIP’

Cisco Unity Connection

Starting with Cisco Unified Communications Manager (CUCM) 6.0 unified communications (UC) deployments with less than 575 phones and less than 500 voice mail boxes have the choice of using Cisco Unified Communications Manager Business Edition (CUCMBE).

CUCMBE includes the CUCM call routing product and the Cisco Unity Connection (CUC) voicemail solution. It runs on one 7828 server that ships with 6GB of DRAM due to the resource requirements of CUCM and CUC. The 6.x version of CUCMBE shipped with the CUC 2.x. The current version of CUCMBE and CUC as of this writing is CUCMBE 7.1(2a) and CUC 7.1(2a).

CUCM and CUC use the same appliance model using the Linux operating system and the IBM Informix database. CUC portion supports up to 16 ports (IP sessions) when running co-resident with CUCM in a CUCMBE solution. CUC in a co-resident configuration supports most of the feature set of standalone CUC with the exception of redundancy and networking solutions. Redundancy for CUCM and CUC in the CUCMBE solution can only be provided by survivable remote site telephony (SRST) running on a Cisco router.

The CUC 7.x standalone platform provides redundancy using a publisher/subscriber database model. The solution provides redundancy and active/active load balancing for up to 2 servers per cluster. The CUC standalone solution is scalable up to 288 ports and 10,000 mailboxes when both servers are active (144 ports per server). CUC can support up to 7,500 users per server when unified messaging capabilities are used with the Cisco Unity Inbox, Cisco GUI clients (VMO or VMN), and IMAP connections.

Larger deployments can take advantage of CUC’s networking capabilities. Up to five clusters can be digitally networked together in the 7.0 version of the product, while 7.1(2a) can support networking of up to 10 clusters. Voice profile for internet mail (VPIM) integration with external directories is supported as well.

Author: Dennis Hartmann

References
Documentation Guide for Cisco Unity Connection Release 7.x

AUC – Administering Cisco Unity Connection v7.0

IUC – Implementing Cisco Unity Connection v7.0

NAT and PAT, Part 2

In Part 1 of this series, we discussed static NAT. While static NAT works, since it uses manually constructed “one-to-one” translations, it’s not scalable. For example, translating all of the legal host addresses on the 10.1.2.0/24 subnet would require 254 lines. And if we were dealing with the entire 10.0.0.0/8 network, covering all possible addresses would require over sixteen million lines! The solution is “dynamic NAT”.

In dynamic NAT, instead of specifying the translations one-by-one, you give the NAT device some rules that specify which addresses are translated to what. In the case of a Cisco router, the addresses to be translated are specified by an access control list (ACL), and the addresses to which they are translated are specified by a “pool”.

For example, to translate any address on the 10.1.2.0/24 subnet (those permitted by ACL 1) to an address on the 200.1.2.0/24 network (as specified by the pool named “Test”), you could do this:

  • Router(config)#ip nat inside source list 1 pool Test

Continue reading ‘NAT and PAT, Part 2′

Voice Gateways

Many small branch deployments of IP telephony require analog interfaces for connectivity to the PSTN, fax machines, security systems, analog phones, and other analog devices. There are many options available that we will explore in this post.

If there is a voice capable Cisco router at the branch office with available voice interface card (VIC) slots, the most economical choice would be to purchase FXO or FXS analog voice interface cards for the router. Foreign exchange station (FXS) ports generate dial tone and the devices that connect to the RJ-11 port expect dial tone. Analog phones and fax machines are good examples of the type of devices that would connect to an FXS port. Foreign exchange office (FXO) ports are ports that expect dial tone from the other side.  FXO ports are normally used for PSTN connectivity. Second generation VIC2 hardware must be used on new integrated services routers (ISR) like the 2800 or 3800 routers. The FXO and FXS modules come in two-port and four-port flavors as follows:

  • VIC2-2FXS
  • VIC2-4FXS
  • VIC2-2FXO
  • VIC2-4FXO

If there are no voice interface card slots available on the router, an analog telephone adaptor (ATA) or voice gateway (VG) device can be used at the branch office, but these devices only provide. The ATA and VG devices are stand-alone devices that provide. Both the ATA-186 and ATA-188 models provide two FXS voice interfaces, but the ATA-188 is end of sale. The voice gateway devices are available in the following form factors providing 2, 4, 24, and 48 FXS port respectively:

  • VG202
  • VG204
  • VG224
  • VG248

References:
Cisco VG 224, VG 204 and VG 202 Analog Phone Gateways

EOS/EOL Announcement for the Cisco ATA 188 Analog Telephone Adaptor

Cisco ATA 186 Analog Telephone Adapter

Cisco VG248 Analog Phone Gateway

Gateways (SRND)

Author: Dennis Hartmann

Seven Pillars of Unified Communications

Unified Communications (UC) may have different meanings depending on who you ask, but the industry has now established what exactly the makeup of UC is, and the minimal applications  that have work together.

All vendors also agree that if enterprise customers fully embrace this concept, it will change the way their employee’s will communicate with each other and their respective customers by being faster and more effective.  This will give corporations the edge they need for reducing costs and drive up their profit margins.

I call it the 7 pillars or application sets that comprises Unified Communications which are:

ucp

Unified Messaging (UM) is the combination of all your fax, email, and voice mail systems into the same messaging infrastructure so that the employees can use their own email client to receive, reply to, or send via that same communications method to others. It speeds up the response time since all employees are tied directly to their email system using their PC or smart phone mobile devices. Additionally, UM should also allow a user to call into their voice mail and have their email read back to them over the phone. And of course this component should also have hands-free operation support for mobile users.

Mobility is demonstrated by having a single number (like an office number) that everyone will normally call me on, however these calls are automatically sent to other phone devices that I might be using as well. For instance, I could be driving into work and have all office calls not only ring at my office desk, but also ring my cell phone at the same time. Since I am still driving into work, I pick the call up on the cell phone, but the person calling me only sees my office number and not my cell phone number. When I reach the office, with a couple of key strokes I can pick the call up on my desk phone. This application should also allow me to call others on any phone I choose, but when the call is received it only shows my office phone number as the calling party. This concept of only giving out one number but having the freedom to have other devices dialed at the same time is quite new to UC.

Instant Messaging is the ability for employees to get quick answers for their questions using text chat. It should also give them the ability to raise the IM chat into an actual phone call by using the same IM software acting as a soft phone device, or by sending a call control request to their office phone to take it off hook and automatically dial the number. Most manufacturers are making their IM client not only function as a text chat platform, but also include soft phone component as well.

Soft Phone is an application that appears to be a phone device on the PC itself. This is a very popular item with mobile IT and sales personnel who tend to travel to multiple locations and want to carry their office with them.

Audio/Video Conferencing is now becoming a “must have” in most corporations. Conferencing deals with having three or more people talking and also sharing presentations (like PowerPoint slides) within a web browser session. Another popular add-on to this would be streaming video so you can actually see the presenter in action.

Applications on the Phone is becoming a very popular feature, especially for conference rooms. With this, a wall phone would display a room conference schedule notifying everyone when a room is supposed to be utilized for official meetings, or the phones could control the conference room lights, projectors, and even the window blinds. My personal favorite is having stock quotes come up on the phone when it has been idle for a period of time.

The last, but most important, UC pillar is Presence located at the heart or center of it all. Presence is defined as the willingness of employees to communicate and the means in which that communication should take place. So, if an employee goes to lunch for 30 minutes but will be available on their cell phone, they would create a presence notification to be displayed on all communication devices that they are at lunch and are reachable by another means.

When you are shopping for a Unified Communications solution, make sure that the vendor you choose can deliver on all the pillars discussed above and can easily handle an expansion when your business starts growing.

Author: Joe Parlas

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Why is a Voice Assessment Important?

After the implementation of a Unified Communication (UC) solution the real work is just beginning. Optimization is the stage where a continuous effort is required to ensure optimal performance from the devices.

As more and more users are added on the system, more and more care is required. It’s similar to starting a new computer for the first time. The PC tends to boot up within seconds and processes information with little delay, but as more applications are installed, the performance begins to degrade overtime. The only way to regain optimal levels is to perform regular maintenance such as defrag the hard drive, add more memory, delete unused files, etc.

This also holds true for a UC solution. As a result, tune-ups are required regularly to ensure the right parts are being adjusted. This process is called a Voice Assessment. It is a process that analyzes the design, servers, firmware, configuration, devices, and system performance. It provides an overview of all the components within the system and the current condition of each component.

I can’t begin to tell you how many companies I’ve seen in the past that have never had a Voice Assessments conducted.  Instead, they choose to take a “break/fix” approach: if it isn’t broken don’t try to make it better, or if it is broken hurry up and fix it.

Voice Assessments discover the underlying problems and issues before they become problems. It is a proactive approach in keeping the system operating at a optimum level at all times.

Voice Assessments are also important as a company grows. For example, as a company grows so does its capacity, which could merit a change in the design.

NAT and PAT, Part 1

In my next few posts, we’re going to discuss NAT and PAT. No, they’re not brother and sister, they’re not even cousins. They are Network Address Translation (NAT) and Port Address Translation (PAT).

It’s common today to use private addressing within an Autonomous System (an “AS” is a collection of routers and subnets under a common administrative domain). Per RFC 1918 (Address Allocation for Private Internets), the private networks are:

  • 10.0.0.0/8 – One class “A” network
  • 172.16.0.0/12 – A block of sixteen class “B” networks
  • 192.168.0.0/16 – A block of 256 class “C” networks

One problem is that per RFC 1918, advertising the address spaces listed above to the public Internet is not allowed. What this means is, that if you send a packet with a “private” source address to the Internet, the destination will not be able to reply to you (because the routers on the Internet backbone won’t know where you are). The solution to this problem is NAT, specified by RFC 1631 (The Network Address Translator).

The first type of NAT we’ll discuss is referred to as “static NAT”. In this method, you build the translation table by hand. For example, let’s say that we want to translate addresses on the 10.1.2.0/24 subnet (private address space) to addresses on the 200.1.2.0/24 network (public). We could translate the first address like this:

  • Router(config)#ip nat inside source static 10.1.2.1 200.1.2.1

The translation tells the router that if a packet with the specified source address (10.1.2.1) hits the inside interface and is bound for the outside interface, translate the source address statically to the second address (200.1.2.1). You can have multiple translation lines, as many as you need, so let’s add some more:

  • Router(config)#ip nat inside source static 10.1.2.2 200.1.2.2
  • Router(config)#ip nat inside source static 10.1.2.3 200.1.2.3
  • Router(config)#ip nat inside source static 10.1.2.4 200.1.2.4
  • Etcetera

The next thing to do is to tell the router which interface (or subinterface) is the “inside” and which is the “outside”. For our example, let’s assume that the FastEthernet0/0 interface connects to our LAN, and the Serial0/0 interface leads to our Internet Service Provider (ISP):

  • Router(config)#interface fa0/0
  • Router(config-if)#ip nat inside
  • Router(config-if)#int s0/0
  • Router(config-if)#ip nat outside

Notice that although we only specified the translation of the source address as the packet transited from the inside to outside interface, the router will automatically translate the destination addresses of packets traversing the router from the outside to inside interface. You can have multiple “inside” and/or “outside” interfaces (or subinterfaces). The beauty of it is that the translation is invisible to all devices, other than the one device performing the translation.

You can view the translation table with the command show ip nat translations, and which interfaces are the “inside” and “outside” (along with other info) with show ip nat statistics.

When you display the translation table (sh ip nat trans), you’ll notice that it specifies “inside local” and “inside global” addresses. The “inside” refers to where the addressed device physically resides (inboard of the “inside” interface, that is, on our side of the router). The “local” or “global” refers to the vantage point from where the address is being viewed. That is, “local” means “as seen from the inside”, and “global” means “as seen from the outside”. In other words, the “inside local” address is our host’s untranslated (actual) address, and the “inside global” address is the translated address (as seen by those outboard of the “outside” interface).

Next time, we’ll examine a variation referred to as “dynamic NAT”.

Author: Al Friebe

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ENUM for the Enterprise

In my last post we looked closely at E.164 and the IETF, and how they made subtle changes to the existing E.164 numbering plan. In essence, all devices with a phone number would reflect the entire number with a plus symbol tagged onto the front of the number, like +14043567893 if you’re in North America. When sent to the provider, the plus symbol (if the provider supports this numbering plan) would tell them that this was an international call and to replace the “+” with that country’s number routine to request international service.

The question is, is this being adopted in the enterprise and if so, how would that make an IP PBX or PBX administrator’s life easier? Continue reading ‘ENUM for the Enterprise’

How to become a voice consultant

With the high unemployment rate and scarcity of jobs, many individuals have given up on job searching and have started their own careers as consultants.Consultants are individuals that have expert knowledge on something of interest. In our case, technology is our area of expertise.

Even though there seem to be a decline in companies hiring Voice Engineers, there is a rise in contract work. This means that Voice Consultants are in demand and more companies are relying on this method to bring in the proper experts, without having the additional expense of hiring someone full-time.  Continue reading ‘How to become a voice consultant’

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