One of the most popular Voice Over IP systems is the Cisco Unified Communication Manager System. Cisco uses several abbreviations for its VOIP system: Cisco Unified (CM), CallManager, and CUCM. This intro will focus on the CUCM architecture; that is, I will explain the most important components and concepts you need to understand if you plan to deploy CallManager in your organization.


At the heart of the system is the CallManager. This device is responsible for the management of the entire system. Management includes:

  • Configuration of end devices, such as phones and voice gateways
  • Call setup and teardown of all calls in and out of the system
  • Configuration of the phone dial plan
  • Configuration of call routing
  • Configuration of trunks

CUCM is built on a Linux server with as little as 1 processor, 8 GB of RAM, and a 120 GB hard drive. This light footprint lends itself well to a VMware environment. Management is provided through a web interface.

CUCM Web Interface

CUCM Web Interface

CUCM devices

Digital phones

Every phone system needs phones! In CUCM, a digital phone is a Linux device running Java and other software to give the end user the traditional desk phone experience. Many models have been offered over the years with basic to advanced features. High-end models like the 82XX series offer color displays and features like built-in webcams. Just like any other computing device, the phones need to be maintained by keeping their firmware up-to-date for security or feature enhancements. Firmware updates are pushed via CUCM.

Cisco 88XX series desk phone

Cisco 88XX series desk phone

Analog phones

CUCM also supports older analog phones. Remember when you (or your parents) had a "home phone," prior to cell phones? That is an analog phone. Analog phones require a voice gateway or an analog telephony adapter (ATA) to connect to the system. Fax machines leverage analog connections to send faxes.

Analog phone

Analog phone

Voice gateways

A voice gateway allows us to connect CUCM to the outside world, aka the public switched telephone network (PSTN), or directly to the Internet, to take advantage of session initiation protocol (SIP). Cisco implements a voice gateway on its advanced services series of routers and via dedicated analog telephony adaptors, or ATAs. The routers use a combination of plug-in hardware cards and software to allow connections to the public phone system.



Cisco 2951 with card slots

Below are some types of connections used by a voice gateway:

POTS—Plain old telephone service (POTS) is an analog service that provides calling capability for one telephone. This is what most homes had in the US until cell phones became the common method of voice-to-voice communication. Individual POTS lines are still used by some businesses that have low calling needs or to run a fax machine for their business.

PRI—Primary Rate Interface is a digital service provided over a T1 circuit that gives a business the ability to make 23 simultaneous calls at one time. These types of lines are used with a PBX system, such as CUCM. While these types of circuits are still available, they are quickly being replaced by SIP.

SIP—Session Initiation Protocol is the replacement for PRI circuits for most businesses. SIP leverages an Internet connection to place phone calls. SIP allows for on-premises and cloud-based phone systems to be implemented and enables businesses to scale as they grow.

Analog Telephony Adapters (ATA)—As mentioned earlier, to connect a traditional analog phone to the CUCM, we use an ATA or a voice gateway. For a small number of analog phones, we would use a two-port ATA. This device is deployed much like a digital phone via an Ethernet connection, and it provides two analog ports to connect a phone or fax device. When many ports are needed, a voice gateway can provide 4, 24, 48, or even 144 analog ports. A common use case for these devices might be a hotel or hospital, where basic phone service is provided to each room.

Cisco 24 port ATA

Cisco 24 port ATA

Trunks—Much like a VLAN trunk in Ethernet, a trunk as it relates to VOIP allows multiple calls to route over the trunk connection. Both the PRI and the SIP are connected to CUCM via a trunk connection.

Other components

Voice Mail—Cisco implements voice mail via a separate server called Unity. A voice mail server can also be used to create a caller attendant integrated voice response (IVR) system. You have likely encountered one of these when you call a business and are greeted with a series of options like "Press 1 for Sales, Press 2 for Marketing," etc.

Collaboration Server—Cisco implements collaboration via its instant messaging (IM) server. Cisco Jabber is an example of an instant messaging and collaboration client. Jabber can be used to collaborate one-on-one or with teams of users. Jabber can also be used to turn your desktop or laptop PC into a "soft phone" to take and place calls.

Emergency Response Server—In many areas of the US and other countries, it is important to identify where someone is physically located when they call 911 or 999. What if they dial 911 because they think they are having a heart attack and later collapse? How does 911 know where the person is located? That is the job of Cisco's Emergency Responder server. It sends location information to EMS teams to use in locating the individual who needs help.

Call Center Server—If you have ever called tech support, waited on hold, and heard a message like "all representatives are currently helping other customers, please continue to hold," you have experienced a call center.

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In this article, we looked at the main components of the Voice over IP system (VOIP) and explained the role each component plays. This was not a complete list of components or features but should give you a good basic knowledge of the components needed to implement a VOIP system.

  1. pepper 1 year ago

    The company I worked for used this VOIP system。CUCM、CUC、CCX、CISCO UCS Server、CISCO 3945 voice router or what else…
    This is definitely the most complex system I’ve ever maintained!!!

  2. Author
    John Kull 12 months ago

    I guess its what you are used to. To someone who has never maintained a voice system it can be overwhelming. I find any complex system is easier to understand when you take it down to its basic parts. That was the goal of the article. I have used CUCM for over 10 years and have also worked with older Lucent \ AT&T that were very cumbersome to maintain. I have found it to be the most flexible system when it comes to integrating with other systems.

    • pepper 12 months ago

      You’re right. I haven’t maintained any voice system before.Although the configuration is complex, it is indeed powerful.

    • Frank Godek 2 months ago

      Good stuff! I did Cisco Call Manager training way back in the early days of Call Manager when it was running on Windows. I’ve never sold a Cisco Call Manager system but I’ve supported many, many VoIP systems over the years and the Cisco training was invaluable. All of the components are basically the same and Cisco’s best practices have adapted well to just about any other system I’ve had to work with. In fact, as a computer/network consultant to a lot of small businesses, when someone has presented something about a VoIP system that doesn’t follow Cisco’s best practices, I’ve learned to be very leery.

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