One of the many things that must be known in order to be a successful CCNA wireless candidate is an understanding of the basics of spread spectrum technologies. This is important because many of the most commonly used wireless technologies in use today take advantage of spread spectrum techniques. This article takes a high-level look at how spread spectrum technologies work and where they are in use in today’s modern networks.

What does Spread Spectrum Mean?

Simply put, Spread Spectrum is the use of a technology that spreads a signal over a frequency spectrum. For example, 802.11b uses the 2.4 GHz band (2.4000–2.4835 GHz) and utilizes channels that are 22 MHz wide with a defined center frequency. The signal is able to be spread across that entire 22 MHz area.

Spread Spectrum Technologies

When dealing with modern wireless networks there are a number of different technologies that are in use. This section takes a look at the different technologies in use on modern wireless LAN networks, specifically 802.11b, 802.11a, 802.11g and 802.11n. In other words, let’s take a look at the different encoding and modulation techniques that are used by these four and how they are used to achieve greater bandwidths.

At the low end of the bandwidth spectrum are the 1 and 2 Mbps options that are available in the 802.11b and 802.11g standards (Originally standardized with 802.11 prime). The encoding method in use at these bandwidths is the Direct-Sequence Spread Spectrum (DSSS) technique using Barker 11 code. The modulation method is Differential Binary Phase-Shift Keying (DBPSK) for 1 Mbps and Differential Quadrature Phase-Shift Keying (DQPSK) for 2 Mbps.

The next step-up in bandwidth was initially defined in the 802.11b standard and is also used with 802.11g, these options include 5.5 and 11 Mbps. To achieve these rates the same two modulation methods (DBPSK and DQPSK) are used but are paired with a different encoding method. To achieve the 5.5 and 11 Mbps rates, the Direct-Sequence Spread Spectrum (DSSS) technique is used with Complementary Code Keying (CCK).

The 802.11a, 802.11g and 802.11n standards utilize the Orthogonal Frequency-Division Multiplexing (OFDM) technique offering a variety of different bandwidth options depending on the modulation technique. 802.11a and 802.11g both utilize OFDM to provide bandwidths from 6 Mbps through 54 Mbps. 6 and 9 Mbps are provided by using Binary Phase-Shift Keying (BPSK), 12 and 18 Mbps are provided by using Quadrature Phase-Shift Keying (QPSK), 24 and 36 Mbps are provided by using 16-Quadrature Amplitude Modulation (16-QAM), and 48 and 54 Mbps are provided using 64-QAM. The difference between the 802.11a and 802.11g implementations is the frequency band being used; the 802.11a standard utilizes the 5 GHz band while the 802.11g standard uses the 2.4 GHz band.

802.11n introduces a number of different features that allow it to not only utilize some of the features of all the other standards, but also reach very high bandwidth potentials. One feature is the ability to utilize 40 MHz channels instead of the 20 MHz channels used by the 802.11a, b and g standards. Unfortunately this can be a blessing and a curse, as the 802.11n standard supports the 2.4 and 5 GHz frequency bands. When using 40 MHz channels, two 20-MHz channels are combined and their combined frequency space is used to provide a single channel. If implementing 40 MHz channels using the 2.4 GHz band, the amount of space and interference can quickly become a large issue, making expansion almost impossible. The other major feature that was introduced in the 802.11n standard is multiple-input multiple-output (MIMO); MIMO provides the ability to utilize multiple spatial streams that can each provide bandwidth numbers equivalent to previous standards. The table below shows the average speeds that are available using a single spatial stream:

Table 1 – 802.11n Speeds and Modulations

There are a number of different 802.11n devices out there that support 2, 3 and 4 spatial streams offering a total theoretical bandwidth of ~600 Mbps with 4 spatial streams.

Summary

There are a number of techniques that can be used to offer various amounts of bandwidth depending on the application; this article simply takes a look at the ones that are used on modern Wireless LAN networks. As the different wireless technologies evolve other new techniques will be developed to squeeze more information into a wireless signal. Hopefully this article gives a good look at what technologies are involved, and offers a starting point for further researching each of these different technologies.

This article takes a look at the changes that have been introduced in the new CCNP ROUTE 642-902 exam providing you with some tips and tricks on how to prepare and pass the challenging exam.

CCNP ROUTE Exam Changes

What we’ll review first are the changes that have been introduced in the ROUTE exam; these include a focus on design and implementation tasks. Previous CCNP exams focused almost exclusively on the technological theory as well as the configuration/troubleshooting Command Line Interface (CLI). A knowledge of the tasks required for both implementation and verification plans is required to pass the newer exams and should be an integrated part of the preparation process.

The hard part is not overthinking what this means; an implementation plan is very simple as it is followed by everyone regardless of whether it is written down or not. An implementation plan is simply a detailed list of tasks that are required for a specific feature to be implemented. For example, how does someone change a flat tire. First, locate spare… second, locate jack and method of tire lug nut removal… third, jack up the car… fourth, remove flat tire… fifth, install spare… sixth, lower car and remove jack… seventh, stow old tire and continue trip.

The same is true for the verification plan, just think of what steps are required to verify that something is fixed or if the problem has been resolved. Using the same flat tire analogy, the person changing the tire would want to make sure that the lug nuts are tight and the tire has sufficient pressure before continuing down the road. When starting to drive down the road on the spare tire the driver would want to ensure that everything “feels right” running on the spare before going on and driving as normal.

CCNP ROUTE Study Direction

So what is the magic bullet for passing the Cisco ROUTE exam? This is simple: Don’t overthink the exam. There are seven high level categories that the ROUTE exam is split into:

1. EIGRP: Enhanced Interior Gateway Routing Protocol
2. OSPF: Open Shortest Path First
3. eBGP: External Border Gateway Protocol
4. IPv6: Internet Protocol version 6
5. IPv6/IPv4 redistribution
6. Layer 3 Path Control
7. Teleworker and branch services

Take each category and focus on it by itself until the material is very familiar. Don’t worry about being familiar with absolutely everything; this is what gets many candidates in trouble. You want to know enough to be comfortable with the material, so that each concept makes sense; you don’t have to know everything that’s written in your CCNP ROUTE book like the back of your hand.

If you don’t have a great deal of Cisco work experience, take advantage of video training (like TrainSignal’s CCNP ROUTE Training or physical classroom instruction. For those candidates with more job experience, the use of video and physical training may not be required. All candidates should take advantage of the self-study materials that exist from a variety of providers; these materials solidify the concepts and provide a continued method of review.

Practice exams in particular are a great resource because they help you not only validate your knowledge before attempting the exam, but also practice taking the exam and getting used to the types of questions that are being asked.

Summary

When it comes down to it, the methods that are used to successfully study for any exam, including the CCNP ROUTE exam, are subjective and highly dependent on the your learning style. However, when it comes down to it, the material is the same and many resources utilize techniques that allow you to learn the concepts regardless of learning style.

Most candidates will find out early in their professional careers which training series/publisher/author/instructor best fits into their specific learning style and will stick with them as long as they provide up-to-date coverage of the new material as it is released. But if studying from a book didn’t go so well last time around try a class or video training instead and see if it works better for you.

The biggest tip is to be well rested the night before the exam and to avoid cramming the night before, this almost always ends up being counterproductive. Give yourself plenty of time to prepare for the exam and make sure that you’re walking into the testing center confident and ready to tackle the exam.

Good luck!

To provide redundancy at the network layer a few approaches can be considered. The most famous protocols used for router redundancy are Cisco’s proprietary HSRP: Hot Standby Routing Protocol and IETF standardized VRRP: Virtual Router Redundancy Protocol. Both protocols have the same concept. They utilize virtual IP addresses shared across several gateways within a network. Only a single gateway at a time can acquire and utilize a virtual address. In case of failure, the virtual address is undertaken by another gateway so that service is never discontinued.

In the past I have described in detail the HSRP protocol. You can refresh your memory and learn more about it in my article on how to achieve network redundancy with HSRP.

In this article we will focus on VRRP which is a standardized protocol used across multivendor routers, although Cisco also supports it. It is the only network layer redundancy protocol that can be used in a network with multivendor routers, so it is very important to get familiar with it.

VRRP Terms

The virtual Router Redundancy Protocol (VRRP) is defined in IETF standard RFC 2338. Before looking into the details of VRRP’s functionality you should get familiar with the following terms related to VVRP:

• VRRP Router: A router that runs VRRP protocol. It may participate in one or more virtual routers.
• Virtual Router: From the Client’s perspective, the virtual router represents the default gateway for hosts within a LAN. It utilizes a Virtual Router Identifier (VRID) within a given LAN subnet and exchanges VRRP protocol messages with other Virtual Routers within the same LAN in order to decide upon the selection of Master and Backup Virtual Routers.
• IP Address Owner: The VRRP router that owns the Virtual Router’s IP address as real interface address and respond’s to clients ARP request for this address.
• Primary IP: VRRP Advertisements are always transmitted using this IP address as source IP address. It is the physical IP address assigned on an interface or VLAN participating in VRRP.
• Master VR: The Virtual Router that is currently elected as master. It is the Virtual Router that serves clients within the specific shared LAN.This VR is the current owner of the Virtual IP address.
• Backup VR: The Virtual Router or set of Virtual Routers that behave as backup routers for the IP address(es) associated with them. The Backup VR immediately takes over the responsibilities of the VR when the Master fails.
• VRID: The Virtual Router Identifier field of the VRRP packet. It has only local significance (within a single LAN) and it is only used for differentiating exchange of messages between Virtual Router instances in a given LAN. It can take a number between 1 and 255.
• Priority: The priority field within the VRRP packet indicates the sending VRRP Router’s priority for the Virtual Router. It can take any value between 0 (which means no participation in VRRP Master election) and 255 (which means that the router owns the IP address associated with the VR). The VR with the highest priority is elected as the Master VR. The default Priority for VRRP routers backing up a VR is 100.

VRRP Message Interaction

One major difference compared to HSRP which is worth telling is the fact that only the VRRP Master VR transmits periodic VRRP messages. This is a major difference compared to HSRP, where, the later specifies that both Master and Backup exchange VRRP messages. We should now examine the VR’s operation on both Master and Backup roles.

VRRP Master

While in Master state, the Virtual Router operates as the default gateway of end-users within the LAN. It responses to ARP requests for the IP address associated with the VR. While in Master state, the VR has to periodically send VRRP Advertisements. The Advertisement Internal is manually configured. By default the advertisement interval is set to 1 second. The Master VR, in case it receives a VRRP Advertisement, it performs the following:

• If the received Priority is greater than the locally configured Priority, transition to the Backup state occurs.
• If the Priority is equal to the local Priority and the IP address of the sender is greater than the local primary IP address, then transition to the Backup state is initialized.

VRRP Backup

While in Backup state, the VR does not participate in any way in normal traffic. It monitors VRRP announcements from the Master and performs the following:

• If an announcement is not received (after a predefined time interval) then, transition to the Master State is performed. To do so, the Backup VR, broadcasts a gratuitous ARP request containing the VR MAC address of the IP address associated with the VR so that layer 2 devices update their forwarding table. From that point onwards, the previously backup VR is now the current master VR.
• By default, if a Backup VR is elected as Master VR and the previously Master (with higher Priority) becomes available, pre-emption takes place, i.e. the active master gives its place to the previous master. Pre-emption can be disabled.

VRRP Message Format

They say that a single picture is equivalent to a thousand words. Well, that is partly true. In our case, I guess, the following picture tells everything about the VRRP packet layout.

Pay attention to the following major characteristics:

• Sender’s source MAC address has the format 00-00-5E-00-01-[XX], where the “XX” consists of a two digit hexadecimal value equivalent to the VRRP Virtual Router Identifier (VRID). For example, a VRRP interface assigned the VRID 12 would have a MAC address of 00-00-5E-00-01-0C.
• Destination MAC address is equivalent the well known multicast address defined for VRRP which is 00-00-5E-00-01-12.

I have included some notes next to the marked items on the above diagram. It is all that you need to know about VRRP message content.

Major VRRP Commands

I would like to close the discussion about VRRP with the major VRRP Interface commands.

Vrrp [VRID] priority [value]
e.g. vrrp 1 priority 110

Vrrp [VRID] timers advertise [msec] [interval]
e.g. vrrp 1 timers advertise msec 500
e.g. vrrp 1 timers advertise 1 …….(seconds)

Vrrp [VRID] ip[ip address]
e.g. vrrp 1 ip 10.10.10.10

No Vrrp [VRID] preempt
e.g no vrrp 1 preempt

In this article we will examine everything you need to know regarding error message logging, reachability and routing troubleshooting as well as technical information collection from Cisco devices. Cisco has incorporated this section into the CCNP TSHOOT exam because it is extremely important to know what your troubleshooting tools can do and how to benefit from them. Learn them now so that you can apply them in real life tomorrow.

Cisco devices are like people; you need to listen to them. They can tell you important things about their hidden thoughts and worries. Always monitor your device logs at frequent intervals. In general, logged messages will assist you in identifying future problems. They will indicate active running malfunctions or even disturbances that happened during your off hours.

Cisco Troubleshooting: Message Logging Levels

The level of message logging is configurable. There are eight distinct levels of logging based on severity. Higher severity messages are given a lower level number. The following table presents these logging levels:

Things to Keep in Mind:

• The highest severity logging level is the “Emergencies” (level 0)
• The lowest severity logs are the “Debug” (level 7)
• Enabling a logging level automatically activates logging of higher severity levels. For example if you configure logging level “3″ then all messages falling into levels zero (0) up to three (3) are logged.

Message Logging Methods

There are four different methods of logging messages in Cisco devices. By default, logging of messages is enabled on the Console and on the device’s internal buffer. The four logging methods are:

• Console
• Internal buffer
• Virtual Terminal ( telnet session)
• Syslog server

The format of the Cisco command to enable logging is:

Logging [method] [level]

The following list displays the commands you need to use to configure each logging method:

• Logging console [level]: This command enables console logging (enabled by default). Use the no logging console command to disable it.
• Logging buffered [level]: This command enables logging of messages to the internal buffer (enabled by default). Use the no logging buffered command to disable it.
• Logging monitor [level]: Use this command to enable logging of messages towards virtual terminal sessions. On your telnet session use the terminal monitor commands to enable the display of messages on your terminal. The command terminal no monitor disables this feature. Also the command no logging monitor disables this logging method.
• Logging [ip address]: This command enables logging of messages towards a syslog server. You can specify several syslog servers by issuing separate commands with the ip address of each syslog server respectively.
• Logging trap [level]: Use this command to specify the level of messages transmitted to the syslog servers. The no logging trap command disables logging of messages to syslog servers.

Display Logging Configuration and Status

To display the configured logging methods and logging messages, issue the show logging privileged executable command. An example is shown below:

Troubleshooting with PING and TRACEROUTE

Do not underestimate the power of the PING and TRACEROUTE commands. You need to know them for your exam preparation as well.

• With the PING command you verify reachability with the remote device. By default, PING sends five ICMP echo requests to the destination IP address expecting to receive an ICMP echo Reply within a time interval of 2 seconds to each request.
• With the TRACEROUTE command you find the path taken to reach a specific destination. It can be used to verify reachability as well. It can provide important information regarding possible network bottlenecks.

Take a look at my article on how to troubleshoot your connections with Ping and Traceroute to learn more.

Important “Show” Cisco Commands

When it comes to identifying hardware problems or service malfunctions, you need to know the basic Cisco commands to use in order to diagnose the problem. Moreover, these are the commands that Cisco experts would ask from you in case you have a maintenance agreement with them, so it is necessary to know them.

When suffering from performance degradation, the following commands are the first to consider:

• Show interfaces
• Show buffers
• Show processes cpu
• Show memory

When you come across IP protocol errors or connectivity errors, the outputs from the following commands need to be evaluated:

• Show ip protocol
• Show ip route
• Show ip interfaces
• Show ip access-lists
• Show ip traffic

There is a single Cisco command that collects a lot of information equivalent to issuing many “show” commands. I am talking about the show tech-support command.

There is another crucial command, a very important one. That is the show version command. This command provides the following important information:

• The installed IOS number and name.
• The system’s Bootstrap and installed BootLoader.
• The system’s uptime.
• The reason for the latest system’s restart.
• The date of the last restart.
• The image filename and stored location.
• Hardware information such as processor type, memory usage, controllers, DSPs, etc.
• The value of the configuration register.

Using Cisco Troubleshooting Tools

Cisco provides a variety of troubleshooting tools to help you identify and isolate potential hardware or software problems. Cisco expects know these tools inside-out. I have presented some of the basic troubleshooting commands in this article, but be sure to learn them well. You will definitely need them!

But, as in real life, “nothing good comes without a price.” Therefore, redundant links may cause frame loops within the network if there is no mechanism to detect these loops. One could ask: What are a few repeated frames within a segment? The answer is that they do not harm the network, but remember broadcast frames occur all the time in switched networks. These frames in bridging loops keep circulating forever. They are exponentially procreating, leading both network bandwidth and resources into starvation.

By the time you notice the problem, it’s too late, your infrastructure is falling down.

Prevent Loops with the Spanning Tree Protocol

IEEE standardized a solution (IEEE 802.1D) to prevent bridging loops in data networks and provide loop-free topologies. This standardized solution is called Spanning Tree Protocol (STP). In this Spanning Tree Protocol tutorial, I will present in simplest terms the operation of STP and indicate how this protocol prevents the creation of bridging loops.

What is Spanning Tree Protocol

As the name implies, STP, spans all switches in a network or subnet. All switches generate and process data messages called Bridge Protocol Data Units (BPDUs). The basic idea behind the exchange of BPDUs is for switches to identify redundant paths and by using the Spanning Tree algorithm, to ensure that there is no loop path in the network.

The STP algorithm is responsible for identifying active redundant links in the network and blocking one of these links, thus preventing possible network loops. The operation of STP is as follows:

• STP enabled switches exchange BPDU messages between them to agree upon the “root bridge;” the process is called Root Bridge Election.
• Once the root bridge is elected, every switch has to determine which of its ports will communicate with the root bridge. Therefore Root Port Election takes place on every network switch.
• Finally, Designated Port Election takes place in order to have only one active path towards every network segment.

Root Bridge Election

Spanning tree enabled switches need to have a common view of the whole network topology. In order to achieve this goal, they communicate between each other using standardized data messages called BPDUs, which are being transmitted using the standardized multicast layer 2 address 01-80-c2-00-00-00. These BPDUs contain various fields.

For the election of the Root Bridge (bridge is equivalent to Switch), the one that will be the initial point of reference, switches manipulate and analyze the Root Bridge ID and Sender Bridge ID fields. Both of these fields consist of a six byte MAC address header and a two byte Bridge Priority header. The switch with the smallest Bridge Priority is automatically elected as the Root Bridge. If Bridge Priority is the same on all switches then the switch with the smaller MAC address is elected as the Root Bridge.

By default all catalyst switches have the same Bridge Priority value (32,768). Let us say that we have three switches as shown in the figure below. All have the same Bridge Priority of 32,768. All switches start by sending BPDUs with a Root Bridge ID and Sender Bridge ID equal of their own. After a few message exchanges, the root election process converges and the Switch with the lower MAC (00-00-00-01-01-01) becomes the Root Bridge.

Learn more about the process of Root Bridge Election in this video from CCIE Chris Bryant: Video: The Root Bridge Election.

Root Port Election

Now that the Root Bridge is elected, every non-root switch has to select a root port, i.e. a port that has the best path towards the Root Bridge. The election of the Root port is determined by the four byte Root path Cost field within each BPDU. Here’s how whole concept is comprised:

• Every switch port has its own path cost based on the port’s bandwidth (equal to 1000Mbps divided by the port bandwidth in Mbps as specified in the original IEEE 802.1D standard).
• The higher the bandwidth, the lower the path cost across the specific port.
• The Path Cost is added to the received Root Path Cost for each BPDU received. Root switch has Root Path Cost of zero (0) for all its ports.
• The port with the lowest resulting Root Path Cost on every non-root switch is finally elected as the Root Port.

Here’s a schematic representation to help clarify this concept.

Learn more about the process of Root Port Election in this video from CCIE Chris Bryant: Video: Root Ports and Designated Ports.

Designated Port Election

The final step of the Spanning Tree Protocol’s computational process is the election of one Designated Port on each network segment. The election of the Designated Port is also based on the Root Path Cost. In case the two or more ports have the same Root Path Cost, the switch with the lower Sender Bridge ID wins and its corresponding port is selected as the segment’s Designated Port.

Any port which is not a Root Port or a Designated Port moves into the Blocking State where it cannot receive nor transmit frames, ensuring that the network is loop-free. Keep in mind that all ports of the Root Bridge are considered Designated Ports and can not be blocked. In our sample network design, the election of the Designated Port on every segment is shown below.

Learn more about the process of Designated Port Election in this video from CCIE Chris Bryant: Video: Root Ports and Designated Ports.

STP Convergence

Traditional Spanning Tree Protocol, by implementation, takes about fifty (50) seconds to adapt and converge to topology changes. In simple words, whenever a topology change occurs in the network (e.g. a link goes down-up), no frame forwarding takes place for about fifty seconds until STP convergences. This is a lot of time of inactivity especially in large networks where topology changes may happen relatively often.

Therefore, great caution needs to be taken where to activate STP. As a rule of thumb STP should be disabled on access ports. To do that you should set all access ports as portfast (meaning that these ports should be put immediately back in forwarding state and avoid the 50 seconds of blackout) and also enable bpdufilter on those ports so that they do not participate in STP.

The necessary commands on interface configuration level, that you need in order to achieve this are:

• Spanning-tree portfast
• Spanning-tree bpdufilter enable

Spanning Tree Protocol Resources

Now that you’ve seen the overview of how you can prevent loops with the Spanning Tree Protocol, continue your learning with these STP Resources:

• Cisco Switching and Spanning Tree Protocol (STP) Basics
• Video: So What Happens if I Turn STP Off?
• title=”STP in Action – STP Examples”>Video: STP in Action – STP Examples
• Video: STP Interface States
• Video: Rapid Spanning Tree Protocol (RSTP)

The installation of the Cisco Unified Communications Manager (CUCM) can seem like a daunting task. This article takes a look at the basic steps required to install the base operating system and the CUCM application on a new device.

Cisco Unified Communications Manager Installation

There are a number of caveats that need to be reviewed before a new CUCM installation. As there is a long list of things to be aware of, the easiest method of review is by reading the sections on these in the Cisco installation guide for the specific version of CUCM that is being installed. This article uses CUCM version 8.6(1) as its source, some steps may be slightly different depending on the version, this installation guide can be found here.

Step 1:

One option that is available is to create an answer file that is used for performing an installation unattended. If this answer file has been already created and copied to a USB key, insert it into the target device.

Step 2:

Insert the installation media into the target device and reboot.

Note: The operating system and CUCM application are installed using the same installation program.

Step 3:

The first thing that is prompted is a media check, this check verifies that the DVD media is intact and has no read problems. If this is the first time the media has been used, performing a media check is cheap insurance that everything will be installed correctly. If the media has been previously verified, this check can be bypassed.

Step 4:

This step checks to ensure that the hardware is correctly configured. Hardware configuration and additional reboots may be required.

Step 5:

The next thing that is prompted is product deployment selection, three options are possible, these include:

• Cisco Unified Communications Manager (CUCM)
• Cisco Unity Connection (CUC)
• Cisco Unified Communications Manager Business Edition 5000

As this article is focused on CUCM, this will be the product to select. Only products that are supported by the current hardware will be selectable.

Step 6:

The next screen is for the Platform Installation Wizard which will prompt to proceed with installation. This screen also offers other methods to proceed including unattended installation and to configure previously installed hardware. Simply select the method that is specific to the installation.

Step 7:

Another secondary step that is available is to upgrade to a more recent service release during the installation. If this is an option that is relevant to this specific installation select yes, the installation will continue on and eventually reboot (at the end of these steps), on reboot an Install Upgrade Retrieval Mechanism Configuration window will display and go through this process.

Step 8:

The next window will prompt for a timezone selection, select the correct timezone for the device.

Step 9:

The installation wizard will then prompt for Ethernet configuration settings including speed and duplex settings. Select either Yes (for autoconfiguration) or No and select the appropriate Ethernet settings.

Step10:

The next window will prompt for MTU configuration, if the default MTU on the network is different than default then enter the appropriate MTU. Typically, the MTU should be left at the default settings, incorrect configuration can affect network performance.

Step 11:

The next window will prompt for device network addressing configuration, the device can be configured to automatically obtain an address through a Dynamic Host Configuration Protocol (DHCP) server or it can be manually configured.

Step 12:

The next window will prompt for the device Domain Naming System (DNS) configuration information, enter the DNS information for this device. Once this step is complete the device will reboot with the configured settings. After reboot the device, the basic installation of the device will be complete and the specific settings will need to be configured depending on the specific application of the device, for example, is the device stand alone or is it part of a cluster.

Once the basic installation process has been completed the user can then configure the device with the settings specific for install. These options include whether the device will be included as part of a CUCM cluster.

Summary

This article reviews the basic steps that must be completed in order to build CUCM. There are a number of different configurations that can then be used to deploy the specific settings required for the specific installation. Hopefully this article can provide a base guideline for those looking to install CUCM in the future.

• Cisco Unified Communications Manager (CUCM)
• Cisco Unified Communications Manager – Express (CME)
• Cisco Unity Connection (CUC)
• Cisco Unity Express (CUE)
• Cisco Unified Presence (CUP)

If you’re thinking of going for your Cisco CCNA Voice Certification then this is a great place to start getting familiar with the different Cisco Unified Communications components.

Cisco Unified Communications Manager

One of the keystone products of Cisco’s Unified Communications products is Cisco Unified Communications Manager (CUCM). CUCM provides a product that enables the unification of voice, video, data and mobile application centralized through a single product. CUCM integrates into all of the other Cisco Unified products enabling support for a very large number of potential Unified Communication solutions. CUCM can be used as an appliance solution using Cisco’s 7800 Media Convergence Servers or Cisco Unified Computing System (UCS) B200 and C210 M2 Rack Mount and Blade Servers as well as installed on a number of different third party servers.

Cisco Unified Communications Manager – Express

The Cisco Unified Communications Manager – Express (CME) provides a scaled down version of CUCM that is able to be deployed on Cisco Integrated Servers Routers (ISR). CME is intended to be deployed in smaller businesses that don’t require the full CUCM solution or an enterprise branch offices that have limited connectivity into the main CUCM solution. CME provides most of the features that would be required in these offices and is a very popular solution in these situations. CME provides the ability to support many features that are conventionally associated with key systems and private branch exchanges (PBX) including IP telephony, voice gateway services, voicemail and auto attendant features (with Cisco Unity Express).

Cisco Unity Connection

Cisco Unity Connection (CUC) is a voice and unified messaging platform that provides the ability to access and manage voice messages in a number of different ways including through email, web browser, IP phone and smartphones. CUC also provides access to a powerful speech engine that is able to not only read text messages but also perform speech recognition. CUC can be installed on Cisco UCS B200 M2 and B210 Rack Mount servers and B200 M2 Blade servers.

Cisco Unity Express

The Cisco Unity Express (CUE) provides a subset of functionality that is provided by CUC in a smaller package that is deployed on Cisco Integrated Servers Routers (ISR). CUE can be integrated into a larger CUC and CUCM solution or implemented with only CME to a small business or office. CUE specifically provides local storage and processing of integrated messaging, voicemail, fax, auto attendant, and interactive voice response (IVR).  CUE can be deployed in a couple of different form factors depending on the series of ISR being used. When being deployed in Cisco 2800 and 3800 series routers, CUE can be deployed using the Cisco Unity Express Network Module (NME-CUE) or Cisco Unity Express Advanced Integration Module (AIM-CUE or AIM2-CUE-K9). When being deployed in Cisco 2900 and 3900 series routers, CUE can be deployed using the Cisco Integrated Services-Ready Engine (ISM-SRE-900-K9) and Service Module Services-Ready Engine (SM-SRE-700, 710, 900, 910-K9).

Cisco Unified Presence

Cisco Unified Presence (CUP) provides an enterprise instant messaging (IM) and network based presence solution that integrates into the Cisco Unified Communications products. CUE provides the ability for clients to support many different features including instant messaging, presence, click to call, phone controls, voice, video, visual voicemail and web collaboration.

Summary

Cisco’s unified communications solutions provide an organization, regardless of size, several options to maintain communications and collaboration. This is vital in modern organizations. This article provides just a brief introduction of the most popular of these options, if deploying this type of solution, take the time to review Cisco’s full product offerings.

Is VoIP a Necessity or Superficiality?

Recently, a private company manager asked me a reasonable question: “Why do I need to change my company’s phone system if I’m happy with it?” Well, if you are happy with your existing phone system or even if you do not have a phone system but you are still happy with your voice connectivity, it’s probably because you haven’t heard about VoIP or seen what it’s capable of. Is there a company manager that does not want to reduce maintenance, management and voice call costs?

It is obvious that investment in VoIP is a necessity not only for new companies but also for existing ones that already have a mature and well served telephony system.

VoIP Connectivity Solutions

Let’s take a closer look at 3 VoIP connectivity solutions that will fit the needs of any company.

• Simple VoIP Connectivity

Small companies or individuals that have one or two telephone lines can easily switch to VoIP without loosing their existing phone numbers. Moreover they can benefit from the concept of virtual numbering and acquire extra numbers within a reasonable cost. Most VoIP providers offer four or even eight telephone numbers through a single connection line which means immediately 50% or even 75% reduction in monthly fixed costs. All you need to do is to choose your VoIP provider and leave the rest to them. They usually install an Integrated Access Device (IAD) at your premise where you can attach your existing telephone equipment or your new VoIP devices and that’s it. Furthermore, most VoIP providers offer attractive call offer schemes like for example free local calls to landline numbers or free minutes of specific international calls. Nevertheless the bottom line is that you reduce both your monthly fixed costs and ongoing voice call expenses.

• Upgrade Traditional PBX Connectivity

For the companies that already possess a traditional telephone system of their own there is a way to keep their existing system and at the same time reduce their costs and enhance their service with VoIP. A VoIP gateway can be used to interconnect the existing telephone system with the VoIP world. There are VoIP providers that offer PSTN or ISDN gateways where customers can connect their existing analogue phones, ISDN phones, analogue PBXs or ISDN PBXs with the VoIP Provider’s network. You existing numbering plan can be maintained and fixed monthly costs and call charges are reduced while at the same time value-added services that the VoIP provider is offering can now be used by the traditional telephony world.

• IP PBX Integration into VoIP Solution

For the demanding users and those that foresee the future, a full IP PBX solution is the ideal solution. I am not talking about a hardware PBX system. I am really talking about a software PBX solution where the PBX software is installed on one of the network servers or just a standalone PC. All users (IP terminals directly and analogue devices through an ATA adapter) are registered on the IP PBX, connected to the VoIP provider’s network through a VoIP trunk connection. More advanced services can be offered, even services that VoIP providers have not yet implemented. The enhanced service perception along with the reduction of running costs are the main benefits of this solution.

5 Benefits of Software PBX for VoIP

I have identified 5 main benefits of using software PBX for VoIP:

1. Cost Reduction: Software IP PBXs are much cheaper than hardware IP PBXs. The installation costs as well as maintenance and upgrade costs are significantly reduced. Software IP PBXs use the open SIP standard, therefore they can interoperate with all major VoIP client manufacturers in contrast with hardware proprietary IP PBXs which interoperate with specific terminal equipment. Therefore, the cost plan depends upon the customers’ selection of terminal equipment.
2. Scalability: Software IP PBXs are powerful in terms of call capacity (Calls per second) and extension support (local terminals or VoIP lines). Therefore, network expansion is not an issue with Software IP PBXs while proprietary hardware systems easily reach their service limits and usually require extra hardware modules to support growing needs.
3. Installation: Software IP PBXs can be easily installed and configured even on Window workstations without the need of having professional expertise to do so. The configuration of VoIP phones as well as software phones installed on personal computers is also an easy to do task.
4. Advanced Customer Service: Because of the software-based nature of the PBX, new features and advanced services can be easily incorporated to deliver better customer service. Not to mention the short time needed to implement a new service and incorporate it into the whole system.
5. Ease of Management: Software PBXs can be managed via a user friendly Graphical User Interface (GUI) which allows system administrators to easily configure new extensions and apply services to extensions without the need of proprietary, difficult to use interfaces. Even if the software PBX does not offer a GUI for management, its file structure and content can be manipulated very easily.

Conclusion

From any perspective you take, VoIP connectivity constitutes a proven necessity for the business and home users. As for the PBXs, it is definitely an ideal solution for the business customers with many extensions or scattered remote branches. Sophisticated services can be delivered to end-users as well as free phone calls within the company premises or between the company’s branches could be achieved.

I personally vote in favor of software IP PBXs; I believe that their scalability and ease of installation are their strongest points. If you want to learn how to install and configure your own software IP PBX, stay tuned. In my next article I will show you step-by-step how to build your own PBX and start reducing your monthly call costs.

However VoIP deployment cannot and should not take place without proper planning and careful network preparation to accommodate VoIP services. First off, you will need to spend some time doing your own investigation. Various VoIP providers offer different service profiles under different network deployments. Some of them offer VoIP on “closed networks”, not through the internet, by utilizing separate virtual connections on customer CPEs while others provide VoIP through internet connections. Also each one of them offers different Service Level Agreements (SLAs) and performance guarantees. Therefore, before even beginning your network preparation, take some time to check these things first.

Today we’ll take a closer look at what aspects should come under careful consideration and what you’ll need to examine so that you can properly prepare your infrastructure for VoIP. What you need to remember is that Voice service is sensitive to delay, delay variation (jitter) and packet loss. The most important aspects that influence the outcome of these parameters are:

• Bandwidth requirements
• Security
• Voice and data traffic separation
• Quality of Service
• Network resilience and high-availability

You really need to understand each one of the above aspects, so take this opportunity and learn about them in this article.

How to Estimate Your Bandwidth Requirements for VoIP

A proper estimation of bandwidth consumption is very important and necessary for proper planning of needed connection trunks to accommodate VoIP traffic. Your bandwidth calculations should be based on the VoIP codec used. If more than one codec is used then you should consider the “worst-case” codec during the busy-hour where the number of concurrent voice calls is about one quarter of all the users in the network.

Let’s take a look at this example:

• Worst-case codec in terms of bandwidth consumption: G.711
• Packetization interval: 20 ms
• Total VoIP users: 100

Using G.711 with a packetization interval of 20 ms, results in bandwidth utilization of about 90 kb/s per voice conversation. To calculate the bandwidth consumption under the above circumstances, you should multiply one quarter of the users (¼ *100) with the 90 kb/s, which results in bandwidth requirement of 2.25 Mb/s during the busy hour.

There are techniques that can further minimize the amount of bandwidth utilization. Voice activity detection (VAD) for example, is known to conserve about 30% of bandwidth by not transmitting packets during silence periods. In our example, if VAD is used the estimated bandwidth calculation would be 1.57 Mb/s.

Security with VoIP

VoIP is susceptible to the same vulnerabilities as the IP data Network. You can read more about network threats and vulnerabilities in my articles on preventing network attacks and dealing with DoS attacks.

Two suggestions I have for you, regarding VoIP security are:

1. Allow specific IP addresses and transport layer ports for both Voice signaling and media with the use of access lists and restrict as much as you can the usable addresses and ports on the network.
2. Use a dedicated internal firewall to monitor the traffic flow and secure your network from application level sophisticated threats

Voice and Data Traffic Separation in VoIP

You should consider separating your data traffic from VoIP traffic. To accomplish this task, you will need to apply dedicated layer 2 VLANs to each traffic category. This way you can achieve traffic classification and you can easily apply different QoS profiles to each traffic category. Besides, layer 2 tagging of packets, Voice packets can also be marked at the network layer. Nowadays, most IP phones support Differentiated-Services bit-marking. Therefore, traffic categorization can be achieved at the very beginning of the voice packet generation. This way, service quality can be maintained end-to-end. Of course, your VoIP provider if not trusting your QoS marking, should at least apply similar traffic categorization and eventually prioritization to your VoIP traffic.

Quality of Service with VoIP

If your network is used to carry both data and voice packets, then you should definitely consider Quality of Service (QoS). For protecting your Voice streams and to prevent data traffic from overwhelming your voice conversations, policing and traffic shaping should be applied. Of course in small networks layer 2 tagging of packets is usually enough to provide highly acceptable level of quality. It is your VoIP provider’s responsibility to apply sophisticated Quality of Service methods to offer better service to you. You can learn more about QoS in my article QoS Using the DiffServ Model.

Network Resilience and High-Availability with VoIP

VoIP service cannot tolerate any kind of interruption. You must make sure that you have adequate uplink physical links to carry all your VoIP traffic even in case you lose one of those links. It’s even better if you have an alternative way to carry your traffic in case of losing your primary path. Hardware malfunction should also be considered. More than one VoIP components should be used where appropriate either in load-sharing mode or even in active-standby mode. Moreover, configuration settings on your network devices should also be upgraded to adhere with VoIP service requirements. For example Spanning Tree should be removed or upgraded to Rapid Spanning tree because of Spanning Tree’s 60 seconds of inactivity during connectivity changes.

An Uninterrupted Power Supply (UPS) unit should also be considered. You do not want your Ethernet switches and VoIP devices to go offline in case of power failure. My suggestion is to use Inline power over Ethernet switches. These switches are capable of powering up all attached devices by delivering power over the unused Ethernet wires. The switches can then be attached on a UPS, the later one being able to provide uninterrupted service for a reasonable amount of time until power failure is corrected.

Are You Ready for VoIP?

Well, these are my recommendations and advices for preparing your network for VoIP. Don’t hesitate to ask anything that you’re not sure about. Your VoIP provider should be happy to assist you on your VoIP journey and help to guide you and provide adequate solutions to fit your needs.

Don’t take anything for granted; be open-minded and willing to try new things and remember that we are talking about a technology that is supposed to be cheaper and more promising. I am certain that you will enjoy every moment of your VoIP journey.

I will end this article with one final piece of advice: don’t spend much on new, sophisticated VoIP terminals because there is no rush in doing so. You can reuse your old terminal equipment with the use of an analogue terminal adapter to accommodate those devices.

Good luck and let me know if you have any questions!

VLAN Example

A VLAN example is illustrated in Figure 1 below. Figure 1 shows a building network example that includes devices for the staff and students of a university. For security purposes, the traffic from individuals working on administrative devices (staff) could be separated from the traffic generated by the academic devices (students).

A method of separating these different devices could be to have them on separate physical networks; however this type of solution can be expensive and inflexible. A better solution would be to create separate VLANs for administrative and academic traffic.

Figure 1 shows four different common areas that exist in a university setting, two of each belong to either the administrative or academic side of the network. The areas that are in the administrative part of the network are separated into VLAN 10; the areas that are in the academic part of the network are separated into VLAN 20. In order for the devices in VLAN 10 to communicate with the devices in VLAN 20, a Layer 3 device (like a router) is required. The Layer 3 device can then be configured to filter the traffic allowed to pass between the two VLAN’s (if any).

VLAN Trunks

Another part of the understanding of VLANs is how they are used between different devices. Without further configuration, the VLAN configuration of a switch is specific to each individual switch. In many smaller deployments, this works out fine as one single switch is deployed for connectivity. However, on larger deployments where there are multiple switches used over a building or campus then the VLAN configuration needs to span multiple switches, this is done with trunks.

Under normal conditions, a switchport is limited to be in a single VLAN; a trunk allows the switchport to support the transport of traffic on multiple VLAN’s. This is accomplished through the use of IEEE 802.1q trunking. When using 802.1q trunking, a tag is inserted into the frame header to identify the VLAN membership; once the frame reaches the destination switch the tag is removed and sent out on all matching VLAN switchports.

Basic VLAN Configuration

The normal range of VLAN numbers used goes from 1 through 1001; the numbers from 1002 through 1005 are reserved for Token Rink and FDDI VLAN’s.  On most switches, including Cisco, the default is VLAN 1 on all switchports. The VLAN range from 1006 through 4094 is also available if extended range VLAN’s are configured.

In order to configure a VLAN on a Cisco switch use the following steps:

Enter global configuration mode

Step 1.              switch#configure terminal

Create or modify an existing VLAN

Step 2.              switch(config)#vlan vlan-id

Configure a VLAN name (optional)

Step 3.              switch(config-vlan)#name name

Another method of creating a VLAN is to configure a switchport into a nonexistent VLAN.  When this is done, the VLAN is automatically created.

In order to configure a switchport into a specific VLAN on a Cisco switch use the following steps:

Enter global configuration mode

Step 1.              switch#configure terminal

Enter interface configuration mode

Step 2.              switch(config)#interface type number

Configure a switchport VLAN

Step 3.              switch(config-if)#switchport access vlan vlan-id

Summary

The configuration of VLAN’s on modern network is common at the access layers of the network; it provides a method of security which is easy to implement and configure. Hopefully this article gives a basic understanding of the concept and how it can be used.

Learn More About VLANs

If you’re interested in learning more about VLANs, check out our article on How to Configure, Verify and Troubleshoot a VLAN and our free video from our Cisco CCNA training covering Virtual LANs and VTP: VLAN Trunking Protocol.

MLS Basics

The concept behind MLS requires an understanding of how routing has worked in the past. Before MLS was available, a packet that needed to be routed would be sent to a route processor. The route processor would read the packet information and determine, in software, the route that the packet must take according to the routing type used on the device (static or dynamic).

While this process was relatively fast, it did require that the packet be decoded in software that slowed the potential speed of packet forwarding down. MLS provides a method for greatly improving the speed of this process by switching the packet in hardware and eliminating much of the work of the route processor.

Packet Rewrite

The next question is how this is possible without the use of a route processor. The route processor is still used to determine the route of traffic, but the subsequent traffic that matches the same traffic flow is switched. Of course, if the traffic was simply switched without alteration of the packet, further routing would be confused as the packet would still have the Layer 2 information from the transmitting device; this is illustrated in the image below:

This issue is resolved through the use of Layer 3 switched packet rewrite. When a packet is Layer 3 switched, certain fields in the IP packet header must be modified in order to make the packet look like it was routed by the device route processor; these included the following five fields:

• Layer 2 (MAC) destination address
• Layer 2 (MAC) source address
• Layer 2 (MAC) checksum
• Layer 3 IP Time to Live (TTL)
• Layer 3 checksum

MLS Flows and the MLS Cache

Traffic routed through an MLS supporting device is monitored to determine traffic flows.  Once these flows are detected, the identifying information is entered into the MLS cache and traffic matching this flow criteria are then subsequently switched.

MLS supports the following unicast flows:

• All traffic to a particular destination
• All traffic from a particular source to a particular destination
• All traffic from a particular source to a particular destination that shares the same protocols and transport-layer information

The format of the MLS cache is determined by a flow mask; the default flow mask depends on the specific equipment used. The main IP flows masks that are supported are (see a similarity?):

• destination-ip – All flows to a specific Layer 3 destination address.
• source-destination-ip – All flows between a specific source and destination address.
• full-flow – All flows between a specific source and destination address, protocol and protocol port.

MLS Configuration

The most common implementation of MLS on modern networks is to use a switch which is essentially a router with switching hardware. In this case, the device is able to utilize the features of both a routing and switching platform. On most of these devices the use of unicast MLS is enabled by default.

Summary

While the use of MLS on many modern switches may be transparent, it is an important feature on a network that requires a high level of performance. For those engineers reading this looking for a good switching solution, ensure that the switches that are selected support MLS functionality as it will greatly increase the performance of the network.

Cisco QoS is typically configured on modern equipment using the Modular QoS Command-line interface (MQC). With MQC, traffic is classified using the class-map and match commands, traffic policy is defined using the policy-map, class and set commands and policies are assigned using the service-policy command.

Another method which can be used is through the use of AutoQoS. AutoQoS takes much of the manual configuration out of the process and creates class and policy maps which are typical of most traffic (as defined by which version of AutoQoS you are using) and sets up policies on the interfaces you specify in a guided setup.

Let’s take a look at what the configuration looks like for both MQC and AutoQoS.

MQC Configuration

Here are a few of the MQC commands mentioned above; keep in mind that there are more match and set commands available, below is just a sampling.

class-map

router(config)#class-map class-map-name

This command is used to create a specific class-map. The class-map-name parameter is used to specify the name of the class-map and can be up to 40 alphanumeric characters.

match protocol

router(config-cmap)#match protocol protocol-name

This command is used to match a specific protocol. The protocol-name parameter is used to specify the protocol name to be matched, there are several which can be used including dhcp, eigrp, h323, http and irc.

match cos

router(config-cmap)#match cos cos-value [cos-value]

This command is used to match a specific Class of Service (CoS) value. The cos-value parameter is used to specify the CoS value carried in the frame to be matched; multiple cos-value’s can be specified in one command.

match dscp

router(config-cmap)#match dscp dscp-value [dscp-value]

This command is used to match a specific Differentiated Services Code Point (DSCP) value. The dscp-value is used to specify the DSCP value carried in the packet to be matched; multiple dscp-value’s can be specified in one command.

policy-map

router(config)#policy-map policy-map-name

This command is used to create a specific policy map. The policy-map-name parameter is used to specify the name of the policy-map and can be up to 40 alphanumeric characters.

class

router(router-pmap)#class {class-name | class-default}

This command is used to link a policy to a specific class-map. The class-name is used to match the class-map-name configured in the class-map command. The class-default parameter is used to specify the default class-map.

set cos

router(config-pmap-c)#set cos cos-value

This command is used to set a specific CoS value. The cos-value parameter is used to specify the Class of Service value which will be set in the frame.

set dscp

router(config-pmap-c)#set dscp dscp-value

This command is used to set a specific DSCP value. The dscp-value is used to specify the DSCP value which will be set in the packet.

service-policy

router(config-if)#service-policy {input | output} policy-map-name

This command is used to link a policy map to an interface The input and output parameters are used to specify in which direction the policy is to be evaluated. The policy-map-name parameter is used to specify the matching policy-map name.

MQC Example

To wrap this up together a bit, the following is a sample configuration which matches all H.323 traffic and gives it a DSCP value of EF (Expedited Forwarding). The configuration will then be configured to be evaluated on traffic coming into an interface.

router(config)#class-map h323

router(config-cmap)#match protocol h323

router(config)#policy-map h323-policy

router(config-pmap)#class h323

router(config-pmap-c)#set dscp EF

router(config-if)#service-policy input h323-policy

AutoQoS Configuration

There are actually two different types of AutoQos: AutoQoS for VoIP (which was the first iteration) and AutoQoS in the Enterprise (which detects the traffic types and builds policy based on this data).

auto qos voip

router(config-if)#auto qos voip

This command is used to install the AutoQoS configuration onto a specific interface.

auto discovery qos

router(config-if)#auto discovery qos

This command is used to start the traffic discovery portion of AutoQoS in the Enterprise. This command should be run for an amount of time to properly detect traffic types before using the next command.

auto qos

router(config-if)#auto qos

This command is used to install AutoQoS in the Enterprise configuration onto a specific interface.

QoS Resources

To learn more about Qos Configuration take a look at this QoS Whitepaper from Cisco and if you’re interested in going as far as the Cisco QOS (642-642) Exam then I’d definitely recommend the Cisco QOS Exam Certification Guide.

Each one of these classes is then configured for a specific level of service which is provided to the traffic. What parameters are set for each service class is configurable and depends greatly on the requirements of the specific traffic.

QoS Basics

There are four major traffic characteristics which are used to classify traffic, these include:

Bandwidth

This is a simple concept; how much total sustained bandwidth is required for the specific traffic type.

Delay

Delay is typically measured from end-to-end and simply records the amount of acceptable delay from source to destination.

Jitter

Jitter is a little more complex in that it defines delay variation. Delay variation is the amount of variation in end-to-end delay which happens from packet to packet. For example, one packet may be delayed so much more than the second packet that the second packet actually gets to the destination before the first.  With data traffic this is not an issue as some reassembly is expected and part of that is reordering the packet. However, with traffic types like voice, getting to the destination out of order lead to voice quality problems.

Loss

Again a simple concept; this is simply the number of packets which are lost from end-to-end.

When classifying traffic it is vital that each traffic type have the correct QoS parameters assigned in order for the network to prioritize correctly.

QoS Models

There are three QoS models which are used in order to provide the correct traffic characteristics. These three models are: Best Effort, Integrated Services and Differentiated Services.

Best Effort

The Best Effort model is rather simple and is the most common on public networks. This model simply gives all traffic the same amount of priority.  All traffic is routed in the same manner and the speed and path of a specific packet is determined by typical destination based networking protocols and equipment.

Integrated Services (IntServ)

The Integrated Services (IntServ) model is considerably more complex. With this model, traffic coming into the network requests a specific traffic class or specific traffic characteristics. For example, if a voice call is trying to be initiated, the phone will typically request a traffic path with specific low (lower) bandwidth, low delay, low jitter and low loss characteristics. These specific traffic characteristics are requested from  each network device from source to destination, if the specific characteristics are available then they are reserved and the traffic is allowed. If the traffic exceeds the specific characteristics setup at the beginning, the networking equipment may drop those packets which do not conform. The Resource Reservation Protocol (RSVP) is typically used to implement IntServ.

Differentiated Service (DiffServ)

The Differentiated Service (DiffServ) model comes at QoS differently; with DiffServ resources are not reserved at the beginning of a traffic flow like with IntServ. DiffServ utilizes classification and marking mechanisms to specify the expected priority that the traffic type expects, this process is typically done at the entry of the network. Each of these devices is then individually configured to react to these specific markings.

Traffic characteristics are maintained through traffic policing and traffic shaping. With traffic policing, traffic with specific markings will be provided a configured service quality (at this device only). If the traffic exceeds the configured amount of service then the device has the ability to drop all non-conforming traffic. With traffic shaping, the traffic is given a little more flexibility and the device attempts to “shape” the traffic into the configured settings. Again, if the traffic is too much out of conformance the device has the ability to drop non-conforming packets.  The DiffServ model is typically deployed over IntServ as it requires a less complex configuration.

Learn more about the DiffServ Model.

Congestion Mechanisms

Another part of this equation is how the network deals with congestion. All of us have been victim to a slow Internet connection from time to time and this is because of network congestion. Simply put, there is more demand for traffic bandwidth then there is supply to service it. How the network deals with these situations is just as important as how it deals with specific traffic service types.

There are two main ways to deal with congestion: Management and Avoidance. These two can also be used with each other. Congestion management looks to deal with congestion after it is already occurring, while congestion avoidance tries to prevent congestion from occurring in the first place.

Congestion management is provided by queue management and gives the ability to queue traffic at a specific point in order to give the equipment time to forward earlier traffic; it also has the ability to skip certain high priority traffic to the front of the queue in order to provide an even higher level of service. There are many queue mechanisms including:

• Weighted Fair Queueing (WFQ)
• First In – First Out (FIFO)
• Class Based – Weighted Fair Queueing (CBWFQ)
• Priority Queueing (PQ)
• Low Latency Queueing (LLQ)

WFQ and FIFO are the most used as they are the default on many interface types.

There are a couple of congestion avoidance mechanisms: Tail Drop and Weighted Random Early Detection (WRED).

Tail Drop is the default with most devices and simply drops all traffic which arrives to a device with full queues. WRED works by selectively dropping packets depending on the average queue size and the priority markings of the traffic. The specifics of the WRED algorithm are outside the scope of this article as it can be quite complex.

All of these technologies had their various advantages but made it complicated to internetwork between any of these technologies. At the same time these technologies were being used, many companies were looking for methods to connect their various offices together and had few good options that were also cost effective.

Some of these solutions included the technologies listed above and others, but most required either configuring VPN’s between various offices or purchasing leased lines (or dedicated paths) between the various offices at a premium.

Tag Switching and Multiprotocol Label Switching

Out of these situations grew the idea of Tag switching which was developed by Cisco and than a formalized standard with Multiprotocol Label Switching (MPLS); when comparing the technologies, the tag and label are synonymous with each other.

The simple idea behind MPLS is to label each packet as it enters a network; the routing across the network is then routed from a label forwarding table or Label Forwarding Instance Based (LFIB). It must be understood however that MPLS still relies on standard routing protocols like OSPF, EIGRP or BGP.

An IP routing table is formed in the same way as it always was but when using MPLS an added layer of label forwarding is provided which can speed the routing of packets (with labels) and provide a method of traffic separation which is not provided by IP without individually setting up VPN’s for all customer’s traffic.

Layer 2 ½ Protocol

When it comes down to it, MPLS is a layer 2 ½ protocol which sits between the layer two protocol (ATM, Frame Relay, Ethernet…) and layer three (IP, IPv6…). It is at this layer in the model that MPLS adds an additional header which dictates the label (or labels) which are attached to each packet.

MPLS also makes use of an additional protocol (technically there are a few) called Label Distribution Protocol (LDP) which is used when a network is not running BGP to distribute the labels which are assigned by the MPLS-enabled routers. I say when not running BGP because BGP has the capability to advertise these labels within the protocol which is not supported natively with current IGP’s (OSPF, EIGRP…).

Setting Up MPLS

When setting up MPLS, the routers within a network that support and perform MPLS functions are called Label Switch Routers (LSR). Each LSR has the ability to do three main things: push, pop or swap labels from a packet. To push a label simply means to add a label to a packet, to pop is to remove a label from a packet and to swap is to remove and add an alternative label to the packet (think how MAC addresses are changed when forwarding Ethernet traffic from switch to switch).

It is possible for a packet to have multiple labels attached which are arranged in a stack and are considered in the order from the most recent label to the least recent label (the label that was pushed most recently will be used for forwarding until it is popped from the MPLS header).

There are three different types of LSR: ingress, egress and intermediate. The ingress LSR is at the edge of an MPLS network and is the first to insert an MPLS header and label on a packet. The egress LSR is at the edge of the network and is the last point before leaving the network and thus removes all of the MPLS labels and header. Both the ingress and egress LSR’s are considered Provider Edge (PE) routers.

The LSRs which exist within the network are called intermediate LSR’s and are responsible for pushing, popping and swapping labels based on the routing with the MPLS network; the intermediate LSR’s are considered Provider (P) routers.

Services Provided by MPLS

The main services which are provided by MPLS are layer two and three VPN’s and Traffic Engineering/Quality of Service (QoS). Layer two VPN’s are also referred to as overlay VPN’s and are implemented on Cisco equipment using Any Transport over MPLS (AToM). Simply put, when using MPLS’s layer two VPN capability the edge connectivity can be any number of different technologies including ATM, Frame Relay and Ethernet. The traffic that comes in these interfaces is then tunneled via MPLS between the ingress and egress PE routers.

From the customers perspective their edge routers are connected via a dedicated circuit. This circuit within the MPLS network is also referred to as a pseudowire. Layer three VPN’s are also possible using MPLS and are referred to as peer-to-peer VPN’s. These differ from layer two VPN’s because the Customer Edge (CE) routers and the PE routers exchange routing information. From the CE perspective they are connected with the PE directly instead of other CE routers. The PE routers keep different customers data protected through the implementation of Virtual Routing/Forwarding (VRF). VRF is run on the PE routers and provides a separate routing table for each customer. Traffic is then routed between the PE routers to other customer sites using MPLS.

MPLS also has the ability to offer Traffic Engineering (TE) and Quality of Service (QoS). When implanting these within MPLS traffic can be reserved across the MPLS network using Resource Reservation Protocol (RSVP). This traffic is inserted into a TE tunnel and has its path specified based on the reservation setup. This same technology is used to give specific types of traffic priority by reserving specific amounts of traffic based on the request.

Getting Started with MPLS

MPLS is a very versatile way of setting up your network whether you’re a provider or a customer. There are certainly many different methods for implanting the various features which can be daunting but with a little research MPLS is not a hard technology to pick up.

There are several resources available which can be referenced for additional information. The following are good places to start:

• MPLS Fundamentals
• MPLS Configuration Examples and TechNotes
• Juniper MPLS (keep in mind that Juniper does have some difference with Cisco and this article is written from the Cisco perspective; however, the main concepts are the same)

Destination based routing systems make it quite hard to change the routing behavior of specific traffic. With PBR, a network engineer has the ability to dictate the routing behavior based on a number of different criteria other than destination network, including source or destination network, source or destination address, source or destination port, protocol, packet size, and packet classification among others.

PBR also has the ability to implement QoS by classifying and marking traffic at the network edge and then using PBR throughout the network to route marked traffic along a specific path.

Why Use Policy Based Routing?

So why would you do this? Well consider a company that has two links between locations, one a high bandwidth, low delay expensive link and the other a low bandwidth, higher delay lower expense link.

Now using traditional routing protocols the higher bandwidth link would get most if not all of the traffic sent across it based on the metric savings obtained by the bandwidth and/or delay (using EIGRP or OSPF) characteristics of the link. PBR would give you the ability to route higher priority traffic over the high bandwidth/low delay link while sending all other traffic over the low bandwidth/high delay link.

This way the traffic which requires the characteristics of the high bandwidth/low delay link would be possible without sending all traffic over the link.

The implementation of PBR is rather simple as well; it is setup to be configured using a match/set process. Traffic which is to be handled by PBR will be matched using an ACL and then have its path or parameters changed using a set command.

PBR Configuration

The first thing that must be done is the configuration of a route map which defines the policy. This is done through the route-map command which is shown here:

router(config)#route-map map-tag {permit | deny} {sequence-number}

The map-tag is simply a name which is used to identify the specific route-map and the sequence-number is used to set the order that route-map statements are evaluated if multiple statements exist.

The second thing that is done is the configuration of a match command which is used to match the specific type(s) of traffic which will be routed using the configured policy. Traffic which does not match any of these commands will be handed off to the destination based routing mechanism. The most common commands which are used as shown here:

router(config-route-map)#match length minimum-length

This command is used to match specific layer 3 packet sizes; this can be used to send packets of various sizes down different paths.

router(config-route-map)#match ip address {access-list-number | access-list-name}

This command is the workhorse of typical PBR configurations; it can be used to match any number of combinations based on a configured access-list.

The third thing that is done is to configure a specific set command, which is used to control the behavior of the matched traffic. The following commands are typically used:

router(config-router-map)#set ip next-hop ip-address

This command is used to specify the IP address of the matched traffics next hop. Make sure that the IP address which is specified in this command is adjacent to the configured router.

router(config-router-map)#set interface interface-type interface-number

This command is used to specify the output interface of the matched traffic.

router(config-router-map)#set ip default next-hop ip-address

This command is used to specify the IP address of the matched traffics next hop, like the set ip next-hop command. However, this command differs from the set ip next-hop command by having a different routing order. When using the set ip next-hop command traffic is policy routed first then passed onto a destination based routing method. When using the set ip default next-hop the destination based routing method is used first then it will be passed to policy routing.

router(config-router-map)#set default interface interface-type interface-number

This command is used to specify a default output interface to send traffic should no explicit route exist.

When using any combination of these commands within a policy the commands are evaluated in the following order:

1. set ip next-hop
2. set interface
3. set ip default next-hop
4. set default interface

PBR is a very powerful tool which can be used to control the specific paths of network traffic, however make sure to only use PBR when it is needed as destination based routing methods work very well by themselves. Like many other features offered on any type of equipment the feature is designed for a specific set of circumstances and should be used for those purposes to maintain efficiency.

The following are a list of links which can be used to reference PBR:

• Cisco Configuration Guide
• Cisco Command Reference

This year has already had some interesting developments with the usage and adoption of IPv6 into the mainstream, not only in Asia and Europe, but surprising updates in the United States as well. Let’s take a look at some of the latest UPv6 implementations and news.

Sprint IPv6 Deployment

Sprint, one of the largest Internet providers across the globe has just completed the first phase of its IPv6 deployment for larger businesses and wholesale partners utilizing its North America Internet backbone.

Sprint plans to continue its implementation of IPv6 over the next few months and by the end of the year, begin rolling out IPv6 support to its remaining domestic points of presence and then start the process of deploying IPv6 to its Europe and Asia backbones.

This development is a major step forward for commercial adoption of IPv6 and I believe will begin a domino effect across the ISP community to deploy and offer IPv6 services. Large providers have limited deployments now, but once a stronger demand for IPv6 grows, larger scale deployments to backbones for commercial use will follow.

Facebook and IPv6

Adoption of IPv6 has now spread to the social networking world. Facebook recently announced at the Google IPv6 Implementers Conference that it was deploying IPv6 to be used with its social networking site.

Facebook stated that its foray into the IPv6 world was “experimental”, but it shows a trend for sites requiring worldwide connectivity. Users across the globe are already using IPv6 and providing a direct access to those users can increase their customer base.

In Facebook’s world of free social networking, advertisements are the real money makers and providing a larger and more global audience can only bring smiles to hungry vendors that advertise their goods and services there.

As sites like Facebook adopt IPv6, other sites like that offer similar services, including online computer games, will slowly but surely follow suit. Internet marketers are also starting to see the larger global reach that IPv6 will bring, but it will take more ISPs offering IPv6 services especially in North America for them to invest and take advantage.

US Federal Government Requires Vendor IPv6 Conformance

The United Stated Federal Government has put into effect a mandate requiring all IT equipment purchased in 2008 or later supports IPv6. As of July 1, 2010, the Federal Government has taken another step farther in enforcing this compliance.

As directed by the Office of Management and Budget (OMB), the Federal Acquisition Regulation (FAR), which is the primary acquisition regulation for the Federal Government, was modified. The modification included a requirement that all IT equipment must be evaluated and certified by an accredited laboratory for IPv6 conformance in accordance to guidelines set forth by the National Institute for Standards and Measures (NIST) and the USGv6 Group.

The step taken by the OMB is a major one. Changing provisions in the FAR to identify key parameters of IT equipment purchases is not a small undertaking. It is now up to the Federal Agencies themselves to implement IPv6 on the equipment that they have purchased. Groups like the Department of Defense and Department of Energy have already deployed IPv6 for some of their networks and some are working on their own rollouts, but there is more work to be done.

IPv6 Education

IPv6 adoption is continuing to rise, slowly but surely. Understanding of IPv6 in the coming years will become a very important skill for IT professionals and those with transition experience will definitely be in high demand.

Along with the well known routing protocols like OSPF and EIGRP there is also some other routing protocols which get less attention. These include BGP and IS-IS.

BGP (Border Gateway Protocol) is used at the core of the modern Internet and focuses on linking large scale networks, IS-IS (Intermediate System to Intermediate System) is a routing protocol which closely resembles OSPF and actually has been around just as long. IS-IS is a protocol which is used as a competitor to OSPF as it is standards based and is supported by multiple vendors.

The commonly used version of IS-IS is called integrated IS-IS which is simply IS-IS being used to route IP traffic; it is important to note that this was not the original design intent with IS-IS.

IS-IS Protocol Basics

Initially the IS-IS protocol was designed to be used to route OSI traffic and used the Connectionless Network Protocol (CLNP) and was used by the Connectionless Network Service (CLNS). The only detail from the original IS-IS implementation which must be understood on today’s networks is CLNS addressing as it is used for IS-IS administration.

IS-IS defines four routing levels from level 0 through 3. On modern implementations only levels 1 and 2 are used. There is a rather simple difference between the two: Level 1 is used for routes within an area and Level 2 is used for routes between areas. Using level 3 would be for routing between Autonomous Systems.

Now IS-IS areas are very similar to those areas used by OSPF with some differences which must be understood. IS-IS does not have a central area requirement as OSPF does, however areas can only be connected via Level 2 configured routers.

Now if you think about this, that means that each router between areas is typically (but not required) a Level 1 and a Level 2 router; this allows the same router to see level 1 internal routes and pass them to another area via the Level 2 component.

The other major difference is that with OSPF, routers are allowed to be in more than one area at the same time to pass routes between those different areas; with IS-IS,  routers are wholly in only one area, the links between the areas become the boundary between the areas.

IS-IS Addressing

One of the most complicated things for most people to grasp with IS-IS is the addressing structure. As most are not familiar with CLNS, some basics must be understood to properly select an addressing structure.

IS-IS uses an ISO style of address which includes an Area, ID (system ID) and a SEL field.

The Area is split into three different parts including the AFI (1 byte), IDI and High Order DSP (HODSP – internal AS Area). When implementing IS-IS the IDI is typically omitted as it is not required with Level 1 and Level 2 routing. The AFI is short for Authority and Format Identifier and are assigned, the most common being 39 (Country code), 47 (International code) and 49 (Private).

The HODSP is used to assign the area used by the routers and is interpreted by Cisco equipment as anything between the AFI and the System ID, so the length is variable.  The ID is a 6 byte (on Cisco equipment) field which is used to specify the system ID of the device being addressed (IS-IS addresses are assigned to the system not the interface).

The SEL field is typically not changed from a value of 00 as this signifies the device; an address with a SEL of 00 is also referred to as a Network Entity Title (NET).

For example, if a router was given the address:

49.0001.1234.5678.9012.00

than 49 would be the AFI,
0001 would be the area,
1234.5678.9012 would be the system ID and
00 would be the SEL,
this address could also be used as a NET address as the SEL is 00.

Basic IS-IS configuration

On Cisco equipment there are some basic configuration parameters which are required to get IS-IS up and running. The commands to get IS-IS up and running on a router include:

1. router isis – This command is used to enable the start of the IS-IS process on the router
2. net address – This command is used to configure the NET address which will be used to identify this router
3. ip router isis – This command is configured on each interface which will participate with IS-IS

By default, a Cisco router will operate in Level-1/Level-2 mode which supports both levels of routing. This makes IS-IS easier to configure but requires additional memory and processor on each device.

If only Level-1 or Level-2 routing is required on a specific device you can enable only the level required on the device by using the is-type [level-1 | level-2] command while in IS-IS router configuration mode.

It is also possible to disable a level of routing on a specific interface by using the isis circuit-type [level-1 | level-2 only] command while in interface configuration mode.

Well because it is an Exterior Gateway Protocol (EGP) as opposed to an Interier Gateway Protocol (IGP) which is what defines the more commonly known protocols including:

• RIP (Routing Information Protocol),
• OSPF (Open Shortest Path First),
• IS-IS (Intermediate System to Intermediate System),
• IGRP (Interior Gateway Routing Protocol),
• EIGRP (Enhanced Interior Gateway Routing Protocol).

As such it operates in a different way from these other protocols.

BGP Basics

The main purpose of BGP is to exchange routing updates like other routing protocols, but BGP typically does not exchange individual network routes (but it technically can), it exchanges summaries of network routes. This is because the typical use of BGP is over very large networks including the Internet.

Without BGP the Internet as we know it would be quite a bit more inefficient. As it is today the Internet BGP routing tables have over 300,000 active forwarding entries and this is with summarization of over 2 billion addresses. Imagine what these tables would be like without summarization.

For those interested, some ISP’s allow the ability to telnet into their edge BGP routers to view the BGP routing tables (Check out ‘route-server.ip.att.net’).

Configuring BGP

In its most basic configuration BGP acts very similarly to a distance vector routing protocol. Each network which is advertised is selected by choosing the shortest path. BGP just uses a path (Autonomous Systems – AS) hop count instead of a device hop count.

For example, BGP works by routing traffic between AS’s, so if Verizon had AS 12345 and AT&T had AS 54321 then traffic destined for the AT&T network would be routed from Verizon to AT&T with a path of (54321). What this means is in order to reach a specific network on the AT&T network, traffic on the Verizon network would have to be routed to AS 54321.

In the following figure I show three AS’s and their corresponding path metrics:

BGP also has loop prevention built in (although this is an open debate); this is implemented with a simple mechanism which disallows routes to be advertised into an AS if the local AS is part of the path metric.

With the example above, only the most basic AS_Path metric is shown, however there are many different path attributes which can be used with BGP to affect path selection along with AS_Path.

The following is a brief list of the available path attributes, in order of path selection preference:

Basic BGP Configuration

The first thing that must be understood is that each BGP device can have both internal and external BGP connections to other devices. Internal BGP connections are within the same AS while external BGP connections are between different AS’s. This is important because the configuration and behavior is slightly different between the two.

eBGP Configuration

At its most basic the configuration of eBGP requires only two commands, these include:

1. router bgp as-number
2. neighbor ip-address remote-as remote-as-number

What makes eBGP configuration obvious from iBGP configuration is that the AS-number which is used in the neighbor command is different than the AS-number configured with the router bgp command.

It must also be known that with eBGP by default there is a direct connection requirement which is enforced by an advertised TTL of 1. Now when configuring BGP using loopback interfaces this can become an issue as the packet actually takes two hops from the remote device to the physical interface and from the physical interface to the loopback interface.

This issue can be resolved by using the neighbor ebgp-multihop command on Cisco equipment.

iBGP Configuration

iBGP configuration is very similar to eBGP configuration but requires a little understanding of iBGP requirements. By default, iBGP requires that all iBGP devices being used are fully meshed (although there are ways of getting around this). This does not however mean that a direct connection is required but that each iBGP peer must neighbor with each other iBGP router.

The following configuration shows that configuration of an iBGP neighbor is the same as with eBGP:

1. router bgp as-number
2. neighbor ip-address remote-as remote-as-number

The other thing that must be understood is how external BGP routes are advertised into iBGP. See the following figure:

In this example, when Level3-2 advertises the eBGP route for the 192.168.128.0/17 network to Level3-1 it will do this with a next-hop of 10.10.10.1 by default. Now if Level3-1 does not have a valid route to the 10.10.10.1 address then it will be unable to route traffic destined for the 192.168.128.0/17 network.

The most common method of resolving this issue is by using the neighbor neighbor-ip-address next-hop-self command. When using this command the local eBGP peer will advertise the next-hop with its own IP address and not the address configured with the BGP neighbor command.

In this case, Level3-2 could be configured with the neighbor 10.100.100.2 next-hop-self command which would advertise the 192.168.128.0/17 network with a next-hop of 10.100.100.1 instead of 10.10.10.1.

Mastering BGP

When it comes down to learning BGP you must prepare for a change in your perception of network routing. BGP is a different beast than the other internal routing protocols and without the ability to separate the two makes learning BGP almost impossible.

For most engineers, including myself, BGP is a good skill to know but it is rarely used unless your job is specific to carrier routing networks. If you are interested in getting more in depth information on BGP check out Wendell Odom’s ROUTE certification guide as it has BGP information and check out the Cisco or Juniper web sites for in depth information; the links for these are listed below.

• CCNP ROUTE – Official Certification Guide
• Cisco’s BGP Page
• Juniper’s Configuring BGP routing

Up until recently there were only a limited number of locations that these practical CCIE exams could be taken, which limited many individuals as it added travel costs to the already high exam fee.

In an effort to give these individuals more of a chance to take the CCIE exam Cisco has decided to take the practical exam on the road. This has been done by developing a mobile CCIE lab program which tries to resolve some of these previous issues with location.

The mobile labs will be scheduled in various locations over the globe and be held as long as the demand in the specific location is high enough to justify Cisco’s expense. Generally the schedule plans for up to 5 days which will be used to hold candidates exams. On each day the lab can accommodate 6 total seats. The mobile labs will also be limited (for the moment) to only CCIE Routing and Switching and CCIE Security.

The cost of taking the CCIE lab using the mobile program will be $1,750 which is a bit more than the typical $1,400 which the lab costs in fixed locations but the savings in travel costs off sets this additional expense.

In order to schedule the CCIE mobile lab go to the CCIE Online page.

Mobile CCIE Lab Schedule

The following is a list of all the expected dates and locations of the mobile CCIE program. All entries listed with an ‘R’ have been confirmed and can be registered for.

Now is a good time to introduce all the components that make Voice over IP a reality. Without them, IP telephony as we know it today wouldn’t exist.

In this article I will mainly focus on introducing the major VoIP components as well as describing the functional characteristics of each one of them.

Different standards provide their own unique specification for each VoIP component. To make things simpler, I will stick to the most important components that are globally acceptable and recognized by everyone. I hope that by the end of your reading, you will have a clear picture of what builds up VoIP.

VoIP Components

The four most important VoIP components are:

• Signaling Gateway Controller
• Media Gateway
• Media Server
• Application Server

  •   Signaling Gateway Controller

As you might remember from my VoIP Essentials article, the Signaling Gateway Controller (SGC) is known as a “called agent” because of its call control function and is also commonly referred to as a “Media Gateway Controller” because of its Media Gateway control function.

The SGC entity has multiple roles. It is the heart of VoIP platform; its main role is to connect the PSTN (public switched telephone network) world with the IP world. To simplify, the main characteristics of the SGC component are:

• Support of Signaling System 7 (SS7) protocol stack which is the PSTN world’s main Signaling protocol suite (sometimes a separate entity called Signaling Gateway is used for this exact purpose).
• Full support of voice call control protocols such as H.323 or SIP which are purely IP signaling protocols.
• Full support of media control protocols such as MGCP or Megaco (H.248) which are used for controlling Media Gateway session connections and parameters.
• Generate Call Detailed Records (CDRs) for billing purposes.
• Provide bandwidth management control through admission control mechanisms, in other words, new sessions are admitted only if the system is able in terms of bandwidth to provide acceptable service to them.
• Support of bandwidth policing mechanisms — with the use of media flow profiles, the Signaling Gateway Controller instructs the Media Gateway to monitor the RTP media flow and apply rate limit policies to aggressive flows. This mechanism also preserves appropriate Quality of Service levels.
• Provisioning media connection — allocating media connection characteristics such as coding and packetization to Media Gateways as well as specific DS0 allocation for the reservation of Media resources.

As you can see, the SGC is the most important component of the whole VoIP structure, therefore, it has to be redundant. Hardware or software malfunction of this component is not tolerant. Also due to it’s multitask and multiprocess behavior, it must be powerful in terms of CPU and memory.

  •   Media Gateway

The Media Gateway’s main role is the transmission of voice packets using the RTP transmission protocol. When the media gateway is used in a converged PSTN/IP network it has extra functions to perform such as packetization, since it uses TDM trunks from the one side and IP trunks on the other.

Let’s examine the Media Gateway’s main functions:

• Support of MGCP or MEGACO for call control under the administration of the Media Gateway Controller.
• Transmission of Voice data using RTP –packetization of data is also applied when TDM trunks are interfacing the Media Gateway.
• Support of T1/E1 trunks for transferring voice in SS7 networks.
• Support of different Compression algorithms for fulfilling the requirements of the call as instructed by the SGC.
• Manage Digital Signal Processing (DSP) resources for ideal service offering.

Some sort of high availability can also be achieved by maintaining redundant IP links. The capabilities of the Media Gateway in terms of concurrent call support, mainly depends on the capacity of onboard DSPs and also the selection of codecs since different codecs have different processing requirements.

  •   Media Server

A Media Server is used where added features are needed such as voicemail or video conferencing. Moreover, a media server is used when special tones or announcements need to be transmitted. Therefore, the media server has an important role within the VoIP architecture.

The main functions of the Media Server are:

• Transmission of call progress tones and special service announcements.
• Voicemail functionality.
• Voice activated dialing.
• Voicemail to email transmission –voicemail can be transmitted as attachment to an email address.
• Support for Interactive Voice Response (IVR) — call routing or even service activation can be performed based on dialed DTMF digits. The caller according to voice menus selects the appropriate DTMF digit that triggers the required service.

The Media Server is mainly controlled by an Application Server using SIP or pure XML. For the proper transmission of IVR, tone and announcement media proper IP routing towards the Media Gateway should exist.

  •   Application Server

The major responsibility of an Application Server is to provide value-added services to the IP network. Global and customer specific services are provisioned here. Call characteristics and session specifications are influenced by the application server.

The main functions of the Application Server component are:

• Support of customized private dialing plans.
• Basic service offering –basic services such as call forward always, call forward on busy, call waiting, call transfer, call park and voicemail are offered though the Application Server.
• Advance service offering — advanced features such as call authorization using PIN, remote office, “follow me” plans can be offered by this component.
• Generation of Call Detailed Records (CDRs).
• FreePhone Service — support of 800 number service where charging is applied to the called party.

The Application Server is the brain of the VoIP architecture. It communicates with the Signaling Gateway Controller through protocols such as H.323 or SIP. Services are implemented here and allocated to customers.

It is very important to have high availability configuration; you can not tolerate service interruption in any way.

More VoIP Information to Come

I tried grouping the most important functions of each VoIP call component and present them to you in the simplest possible way. I hope you found this information useful.

Stay tuned; more VoIP articles are yet to come.