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* Assists traditional telco engineers and technicians in designing, implementing, and supporting VoIP networks
* Provides a step-by-step approach to designing a VoIP network
* Outlines each VoIP technology components with detailed descriptions of possible applications
* Provides a thorough description of Voice over IP for those new to the technology

Voice over IP (VoIP) is a fast-growing technology within telcos and ISPs. VoIP is significantly reducing the cost of transmitting telephone calls and enabling small companies to enter the +$50 billion marketplace. Many telcos and ISPs are entering this market while VoIP is still in its infancy. As a result, there is an urgent need for telco and ISP engineers to learn this new technology. This book will assist telco engineers to transition to IP networking with a focus on VoIP networks. This book introduces the fundamentals of IP networking as it applies to transmitting voice calls using IP packets. Key telco-based technologies such as SS7 are reviewed. A unique approach is that the book outlines each component of VoIP technology while describing the application of the technology. This is accomplished by describing key steps in designing an operational VoIP network, thereby enabling the reader to understand how this technology is applied.

Chapter 2, “VoIP Network Architectures: H.323, SIP, and MGCP,” describes a managed voice and data service architecture that uses a call agent. A call agent solution is commonly used for the small and medium-sized business (SMB) market. This solution uses an integrated access device (IAD) at the customer premises that supports voice and data using IP or Asynchronous Transfer Mode adaptation layer 2 (AAL2) transport over a T1 access link to the service provider.

This chapter continues the focus on offering managed services to SMBs. Service providers that traditionally use time-division multiplexing (TDM) access and that need to add other service offerings can bundle voice and data services over a single access link to the customer. For example, traditional T1 circuits that are offered to customers to interconnect their private branch exchange (PBX) to interexchange carriers (IXCs) can now be used instead for integrated voice and data traffic, thereby eliminating the need for multiple access links between the customer and the service provider.

Three areas are discussed to help provide an overview of bundled voice and data service architectures:

*

Overview of Managed Voice and Data Services
*

Managed Voice and Data Services Using AAL2
*

Fundamentals of AAL2

Overview of Managed Voice and Data Services

Integrated voice and data is a new service that is offered by service providers. The architecture design to deploy managed voice and data services is based on many factors. One of these factors is the required customer premises equipment (CPE), which is based on the type of business customer.

Two general types of business customers exist. One is a business customer with fewer than 100 users, such as a doctor’s office, an insurance agent’s office, or a small home office. These businesses are normally single-site locations that require telephony services, Internet access, firewall, and Virtual Private Network (VPN) services. Typically, these businesses do not have older networking protocols, such as AppleTalk or IPX, and they do not have a full-time support staff to maintain their own private network. A service provider can support these services with an IAD on the customer premises, such as a Cisco 2400.

The second type of business customer is an enterprise customer. An enterprise customer has a large installed base of devices that supports many flavors of protocols, sophisticated routing designs, multiple T1s, and back-hauling needs. An enterprise customer needs a multiservice platform, such as the Cisco 2600 and 3600. Many of these enterprise businesses have their own large private networks and their own full-time staff to maintain their multiservice network. However, because of various reasons, such as fast growth and economics, many of these large customers are outsourcing some or all of their services to service providers.
Integrated Access Architectures

Traditionally, service providers offer TDM services that connect a customer’s PBX to an IXC Class 4 switch, which provides long distance voice services. Many of these service providers are currently switching from using a TDM-based infrastructure to using a packet-based infrastructure, either IP or ATM. This approach allows for a more efficient method to provide voice transport and also helps to integrate voice and data services over one access link to the customer premises.

NOTE

Many incumbent carriers are adding IP to their core ATM network by inserting Multiprotocol Label Switching (MPLS) technology. This change enables service providers to shift from transport service offerings to IP-based service offerings.

ADSL and T1 ATM are two types of access technologies that can be supported between the SMB and the service provider.
Other Cisco IADs

Other IADs available for SMBs are the Cisco 827-4V and the 1750. Both of these IADs support four Foreign Exchange Station (FXS) interfaces, and the 1750 also supports FX0 and E&M interfaces. The 827-4V supports a fixed configuration that includes Ethernet and DSL WAN-access only, and the 1750 supports a modular configuration that includes Fast Ethernet and multiple WAN options, such as DSL and T1 access. To help further differentiate these two IADs, the 1750 supports dual WAN interfaces for WAN backup or load sharing, hardware Triple Data Encryption Standard (3DES) encryption, and Open Shortest Path First (OSPF) and Border Gateway Protocol (BGP) routing—these capabilities are not present in the 827-4V. The 827-4V is well suited for small offices that do not require the extra capabilities of the 1750 and are not concerned with expandability, but do require core services such as basic voice (FXS), VPN, and a firewall from their service provider.

More info:Voice-Enabling the Data Network: H.323, MGCP, SIP, QoS, SLAs, and Security (Networking Technology)

# Understand the basics of enterprise and public telephony networking, IP networking, and how voice is transported over IP networks
# Learn the various caveats of converging voice and data networks
# Examine the basic VoIP signaling protocols (H.323, MGCP/H.248, SIP) and primary legacy voice signaling protocols (ISDN, C7/SS7)
# Explore how VoIP can run the same applications as the existing telephony system but in a more cost-efficient and scalable manner
# Delve into such VoIP topics as jitter, latency, packet loss, codecs, QoS tools, and security

Voice over IP (VoIP) has become an important factor in network communications, promising lower operational costs, greater flexibility, and a variety of enhanced applications. To help you understand VoIP networks, Voice over IP Fundamentals provides a thorough introduction to the basics of VoIP.

Voice over IP Fundamentals explains how a basic IP telephony infrastructure is built and works today, major concepts concerning voice and data networking, and transmission of voice over data networks. You’ll learn how voice is signaled through legacy telephone networks, how IP signaling protocols are used to interoperate with current telephony systems, and how to ensure good voice quality using quality of service (QoS).

Even though Voice over IP Fundamentals is written for anyone seeking to understand how to use IP to transport voice, its target audience comprises both voice and data networking professionals. In the past, professionals working in voice and data networking did not have to understand each other’s roles. However, in this world of time-division multiplexing (TDM) and IP convergence, it is important to understand how these technologies work together. Voice over IP Fundamentals explains all the details so that voice experts can understand data networking and data experts can understand voice networking.

The second edition of this best-selling book includes new chapters on the importance of billing and mediation in a VoIP network, security, and the common types of threats inherent when packet voice environments, public switched telephone networks (PSTN), and VoIP interoperate. It also explains enterprise and service-provider applications and services.

Article Information
Contents

1. Delay/Latency
2. Jitter
3. Pulse Code Modulation
4. Voice Compression
5. Echo
6. Packet Loss
7. Voice Activity Detection
8. Digital-to-Analog Conversion
9. Tandem Encoding
10. Transport Protocols
11. Dial-Plan Design
12. End Office Switch Call-Flow Versus IP Phone Call
13. Summary
14. References

Article Description
This chapter explains many of the issues facing Voice over IP (VoIP) and ways in which Cisco addresses these issues.
From the Book
Voice over IP Fundamentals, 2nd Edition

Voice over IP Fundamentals, 2nd Edition

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To create a proper network design, it is important to know all the caveats and inner workings of networking technology. This chapter explains many of the issues facing Voice over IP (VoIP) and ways in which Cisco addresses these issues.

Communications via the Public Switched Telephone Network (PSTN) has its own set of problems, which are covered in Chapter 1, “Overview of the PSTN and Comparisons to Voice over IP,” and Chapter 2, “Enterprise Telephony Today.” VoIP technology has many similar issues and a whole batch of additional ones. This chapter details these various issues and explains how they can affect packet networks.

The following issues are covered in this chapter:

* Delay/latency
* Jitter
* Pulse Code Modulation (PCM)
* Voice compression
* Echo
* Packet loss
* Voice activity detection
* Digital-to-analog conversion
* Tandem encoding
* Transport protocols
* Dial-plan design

Delay/Latency

VoIP delay or latency is characterized as the amount of time it takes for speech to exit the speaker’s mouth and reach the listener’s ear.

Three types of delay are inherent in today’s telephony networks: propagation delay, serialization delay, and handling delay. Propagation delay is caused by the length a signal must travel via light in fiber or electrical impulse in copper-based networks. Handling delay—also called processing delay—defines many different causes of delay (actual packetization, compression, and packet switching) and is caused by devices that forward the frame through the network.

Serialization delay is the amount of time it takes to actually place a bit or byte onto an interface. Serialization delay is not covered in depth in this book because its influence on delay is relatively minimal.
Propagation Delay

Light travels through a vacuum at a speed of 186, 000 miles per second, and electrons travel through copper or fiber at approximately 125, 000 miles per second. A fiber network stretching halfway around the world (13, 000 miles) induces a one-way delay of about 70 milliseconds (70 ms). Although this delay is almost imperceptible to the human ear, propagation delays in conjunction with handling delays can cause noticeable speech degradation.
Handling Delay

As mentioned previously, devices that forward the frame through the network cause handling delay. Handling delays can impact traditional phone networks, but these delays are a larger issue in packetized environments. The following paragraphs discuss the different handling delays and how they affect voice quality.

In the Cisco IOS VoIP product, the Digital Signal Processor (DSP) generates a speech sample every 10 ms when using G.729. Two of these speech samples (both with 10 ms of delay) are then placed within one packet. The packet delay is, therefore, 20 ms. An initial look-ahead of 5 ms occurs when using G.729, giving an initial delay of 25 ms for the first speech frame.

Vendors can decide how many speech samples they want to send in one packet. Because G.729 uses 10 ms speech samples, each increase in samples per frame raises the delay by 10 ms. In fact, Cisco IOS enables users to choose how many samples to put into each frame.

Cisco gave DSP much of the responsibility for framing and forming packets to keep router/ gateway overhead low. The Real-Time Transport Protocol (RTP) header, for example, is placed on the frame in the DSP instead of giving the router that task.
Queuing Delay

A packet-based network experiences delay for other reasons. Two of these are the time necessary to move the actual packet to the output queue (packet switching) and queuing delay.

When packets are held in a queue because of congestion on an outbound interface, the result is queuing delay. Queuing delay occurs when more packets are sent out than the interface can handle at a given interval.

The actual queuing delay of the output queue is another cause of delay. You should keep this factor to less than 10 ms whenever you can by using whatever queuing methods are optimal for your network. This subject is covered in greater detail in Chapter 8, “Quality of Service.”

The International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.114 recommendation specifies that for good voice quality, no more than 150 ms of one-way, end-to-end delay should occur, as shown in Figure 7-1. With the Cisco VoIP implementation, two routers with minimal network delay (back to back) use only about 60 ms of end-to-end delay. This leaves up to 90 ms of network delay to move the IP packet from source to destination.
vf330701.gif

Figure 7-1 End-to-End Delay

As shown in Figure 7-1, some forms of delay are longer, although accepted, because no other alternatives exist. In satellite transmission, for example, it takes approximately 250 ms for a transmission to reach the satellite, and another 250 ms for it to come back down to Earth. This results in a total delay of 500 ms. Although the ITU-T recommendation notes that this is outside the acceptable range of voice quality, many conversations occur every day over satellite links. As such, voice quality is often defined as what users will accept and use.

In an unmanaged, congested network, queuing delay can add up to two seconds of delay (or result in the packet being dropped). This lengthy period of delay is unacceptable in almost any voice network. Queuing delay is only one component of end-to-end delay. Another way end-to-end delay is affected is through jitter.

more info:Voice over IP Fundamentals (2nd Edition) (Fundamentals)
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* Master troubleshooting techniques and methodologies for all parts of a Cisco IP Telephony solution-Cisco CallManager, IP phones, gateways, applications, and more
* Learn how to investigate and resolve voice quality problems, including delayed audio, choppy or garbled audio, static and noise, one-way or no-way audio, and echo
* Read about the variety of trouble-shooting tools at your disposal and how and when to use them based on the problem type
* Discover the potential causes of common problems and how to efficiently troubleshoot them to resolution
* Learn how to identify and resolve gateway problems by breaking the components into logical groups and following a methodical troubleshooting approach
* Use best practices recommendations to build a stronger IP telephony deployment and avoid common mistakes

IP telephony represents the future of telecommunications: a converged data and voice infrastructure boasting greater flexibility and more cost-effective scalability than traditional telephony. The ability to troubleshoot an IP telephony environment and the underlying network infrastructure is vitally important, just as it is in any complex system.

Troubleshooting Cisco IP Telephony teaches the troubleshooting skills necessary to identify and resolve problems in an IP telephony solution. This book provides comprehensive coverage of all parts of a Cisco IP Telephony (CIPT) solution, including CallManager, IP phones, gateways, analog devices, database and directory replication, call routing, voice mail, applications, network infrastructure, and more. You’ll learn how to read trace files, determine when to turn on tracing and Cisco IOS(r) Software voice debugging, and how to troubleshoot voice quality issues.

Troubleshooting Cisco IP Telephony shows you how to break down problems to find the root cause. Descriptions of each part of the CIPT solution help you understand the functionality of each part of the solution and how each part interacts with other parts of the solution. You’ll then learn what steps to take and tools to use to identify and resolve the cause of the problem.

t’s 5:30 a.m. on a Monday and your pager goes off. You recognize the phone number— it’s your CEO’s administrative assistant. As the administrator of the company’s 8000-phone IP Telephony network, you assume there’s a big problem. You rush into work and find the CEO’s administrative assistant, who states that several calls for the CEO have been disconnected in the middle of the call, including a call from a very important customer. Where do you start?

Troubleshooting a Cisco IP Telephony network can be a daunting task. Rather than describing step-by-step how to solve specific problems (subsequent chapters provide that information), this chapter focuses on teaching a good troubleshooting methodology: learning how to find clues and track down your “suspect” by breaking the problem into smaller pieces and tackling each piece individually.

A typical IP Telephony network consists of—at the very least—one or more of the following components:

* Cisco CallManager servers
* IP phones
* Voice gateways

These components are in addition to the data network infrastructure that supports voice over IP (VoIP) traffic. More-complex installations can have dozens of servers for different services and redundancy, each server running a variety of applications, as well as hundreds or thousands of IP phones and a large number of voice gateways.

Before exploring the myriad of tools, traces, and techniques available to you that aid in troubleshooting, you must develop a systematic method by which you can focus on the problem and narrow it down until you determine the root cause.

In addition to the information in this book, you should become familiar with the various standard protocols that are used in an IP Telephony network, such as the following:

*

H.323
*

Media Gateway Control Protocol (MGCP)
*

Telephony Application Programming Interface/Java Telephony Application Programming Interface (TAPI/JTAPI)

You should also become familiar with the protocols used when interfacing with the traditional time-division multiplexing (TDM)-based Public Switched Telephone Network (PSTN), such as the following:

*

Q.931 (an ISDN protocol)
*

T1- or E1-Channel Associated Signaling (T1-CAS or E1-CAS)
*

Foreign Exchange Office (FXO)
*

Foreign Exchange Station (FXS)

Additionally, because an IP Telephony network runs over a data network, it is important to understand the protocols that transport VoIP data, such as the following:

*

Internet Protocol (IP)
*

Transmission Control Protocol (TCP)
*

User Datagram Protocol (UDP)
*

Real-Time Transport Protocol (RTP)

Later chapters cover some of these concepts. However, each of the mentioned protocols could take up an entire book on its own, so you should refer to the specifications and RFCs or to other materials that go into detail about these protocols. Appendix A, “Cisco IP Telephony Protocol and Codec Information and References,” provides references to where you can find additional information for each protocol discussed in this book.

On the other hand, because the Skinny Client Control Protocol (SCCP or Skinny protocol, the Cisco-developed protocol that Cisco IP Phones use) is not the product of an industry-wide standards body, this book goes into additional detail about how this protocol works. Understanding the Skinny protocol is essential to understanding how the phone operates and how to troubleshoot problems with it. The Skinny protocol is covered in greater detail in Chapter 5, “IP Phones.”
Developing a Troubleshooting Methodology or Approach

To track down a problem and resolve it quickly, you must assume the role of detective. First, you need to look for as many clues as you can find. Some clues lead you to additional clues, and others lead you to a dead end. As soon as you’ve got all the clues, you need to try to make sense of them and come up with a solution. This book shows you where to look for these clues and track down the problem while trying to avoid as many dead ends as possible.

Troubleshooting a problem can be broken down into two stages: data gathering and data analysis, although your analysis might lead you to collect additional data. The following list is a general guide for steps to take when troubleshooting an IP Telephony problem:

Step 1

Gather data about the problem:

(a)

Identify and isolate the problem.

(b)

Use topology information to isolate the problem.

(c)

Gather information from the end users.

(d)

Determine the problem’s timeframe.

Step 2

Analyze the data you collected about the problem:

(a)

Use deductive reasoning to narrow the list of possible causes.

(b)

Verify IP network integrity.

(c)

Determine the proper troubleshooting tool(s), and use them to find the root cause.

Production Versus Nonproduction Outages

Troubleshooting a problem can occur in one of two timeframes:

*

During a scheduled outage window, such as when you’re installing a new system, adding components, or upgrading for new features or functionality
*

During production hours when the problem affects end users or service

Although the methodology to troubleshoot problems in either of these two situations is similar, the focus on how to resolve the problem should be different. In the case of a service-affecting problem during production hours, the focus should be to quickly restore service by either resolving the problem or finding a suitable workaround.

In contrast, when a problem is found during a new install or scheduled outage window, the focus should be on determining the root cause to ensure the problem is completely diagnosed and resolved so that it does not have the potential to become service-affecting.

For example, if users are encountering a delayed dial tone or sluggish behavior on their phones, you might discover that a high-level process on CallManager is consuming 100 percent of the CPU on one of the servers. During a new install or scheduled outage window, it’s a good idea to investigate what is causing the CPU consumption to ensure that the problem does not return during production hours.

However, if this problem occurs during production hours, the best approach is to stop or restart the offending process and let the redundant systems take over to quickly restore service. After you restore service, perform a root-cause analysis to try to determine why that process was consuming the CPU. The downside of this approach is that you might not be able to further troubleshoot the problem when the process is restarted. Fortunately, CallManager provides many diagnostic traces (if they are enabled prior to the problem) that you can reference after a problem has occurred to see what was happening on CallManager at the time of the problem.

Note that although 100 percent CPU of a high-level process can cause sluggish behavior or delayed dial tone, do not infer from this that 100 percent CPU is necessarily always a bad thing. As of CallManager 3.3(1), low priority tasks (such as phone registrations) can con-sume 100 percent CPU without causing adverse effects to the ability to place or receive calls. Look at the 100 percent CPU as a possible symptom but not necessarily the root cause. In this case, you observe the symptoms of sluggish or delayed dial tone and 100 percent CPU utilization and make a correlation between the two.

If you encounter an event where you are unable to determine the root cause due to insuf-ficient information, it is a good idea to turn on the appropriate traces to ensure that if the problem reoccurs, you will have enough data to identify the root cause.

Sometimes, several service-affecting problems occur simultaneously. In fact, this is not uncommon, because multiple problems often manifest themselves as symptoms of the same root cause. When multiple problems occur simultaneously, focus on the problem that has the greatest impact on users. For example, if some users are reporting dropped calls and others are reporting occasional echo, the two problems are probably unrelated. Trouble-shoot the dropped-call problem first because keeping calls connected is more critical than removing the occasional echo on an active call.
Step 1: Gathering Data About the Problem

So you’ve just installed a new IP Telephony network, or you’ve been given the task of maintaining one—or maybe you’ve taken your first CallManager out of the box and are having problems getting it to run. You’ve encountered a problem. The first thing to do is gather as much information about the problem as possible.
Identifying and Isolating the Problem

Half the battle in troubleshooting a problem is determining which piece of the puzzle is the source of the problem. With so many different pieces composing an IP Telephony network, the first step is to isolate the problem and, if multiple problems are being reported, deter-mine which of the problems might be related to each other and which should be identified as separate problems.

You must also determine which parts of the problem are symptoms and which are the root cause of the problem. For example, if a user complains of a phone resetting itself, it might seem logical to first assume that something is wrong with the phone. However, the problem might lie with CallManager or one of the many routers and switches that make up the un-derlying data network. So although the symptom is a phone reset, the root cause could be a WAN network outage or CallManager failure. You must always remember to look at the big picture when searching for the root cause and not let the symptoms of the problem lead you in the wrong direction. To help you visualize the big picture, detailed topology information is essential.
Using Topology Information to Isolate the Problem

You can take many proactive steps to help make the troubleshooting process easier. One of the first lines of defense is possessing current topology information. One of the most im-portant pieces of topology information is a detailed network diagram (usually created using Microsoft Visio or a similar application). The network diagram should include network addressing information and the names of all the devices. It should also clearly show how the devices are interconnected and the port numbers being used for these interconnections. This information will prove invaluable when you try to isolate which components are involved in a particular problem.

For medium- to larger-sized networks, you should have a high-level overview topology that gives you a general idea of how things are connected and then several more-detailed dia-grams for each piece of the network that drill down to the interface level on your network devices.

Figure 1-1 shows a typical high-level topology diagram for a large enterprise IP Telephony network. Notice that device names and IP addresses are listed in the diagram. This makes troubleshooting easier by allowing you to quickly look up devices to access them. Because Figure 1-1 is a high-level diagram, it does not get down to the interface level of each device.

Most networks are not as large as the one shown in Figure 1-1. However, no matter the size of your network, a similar topology diagram is very useful for quickly sharing information about your network with others who might be assisting you in troubleshooting.

In addition to the network diagram, you should use some method to store information such as IP address assignments, device names, password information, and so on. For a small network, you can use something as simple as a spreadsheet or even a plain text file. For larger deployments, some kind of database or network management application such as CiscoWorks is recommended. Many customers keep all this topology information on a web server as well, making it quickly and easily accessible to others when it is needed the most. Be sure to keep this information in a secure location.

You also need documentation of your dial plan. Some deployments, especially those heavily utilizing toll-bypass, have very complex dial plans. Knowing where a call is supposed to go just by knowing the phone number and from where it is dialed helps you quickly understand a problem.

Figure 1-1Figure 1-1 Sample High-Level Topology Diagram

When your topology information is complete, it should include all the following information:

*

Interconnection information for all devices, including device names and port num-bers. If any patch panels exist between devices, the port numbers should be listed.
*

IP addressing for all network devices (routers, switches, and so on)
*

IP addressing for all telephony and application servers and voice gateways (including data application servers)
*

IP addressing for endpoints (that is, scopes of a DHCP pool)
*

WAN and PSTN service provider names and Circuit IDs for each circuit
*

Spanning-tree topology, including root bridges for all VLANs and which ports should be forwarding and blocking
*

Dial plan information
*

Software version information for all devices

If you are troubleshooting a network you didn’t design, topology is one of the first pieces of information you should obtain, if it’s available. If a topology drawing is not available, it is a good idea to spend time obtaining this information from someone who is familiar with the network and then making a quick sketch. A general topological understanding of the network or at least the piece of the network in question helps when you’re trying to dif-ferentiate the problem from its symptoms. It’s necessary when you’re trying to isolate the problem to a particular part of the network.

For example, if a user reports hearing choppy audio when making a conference call, it is essential to know exactly where in the network the conference bridge device is located in relation to the user’s phone, including all the intermediate network devices. Without a net-work diagram, finding this information could waste precious time. Assume that the network you are troubleshooting looks like Figure 1-1. If the user’s phone is connected to Access Switch 1A, the other conference participants are on Access Switch 1Z, and the conference bridge device is on Voice Switch 1A, you can see that the number of devices is greatly reduced from 100 or more switches and routers to four or five.

What is worse than not having topology information? Having incorrect topology infor-mation can lead to countless hours heading down the wrong path. If you’re going to keep topology information (highly recommended), make sure you keep it current.

Use all the topology information you have to narrow down which pieces of the network might be involved in the problem you are trying to troubleshoot. To further isolate the problem, interview the end users who reported the problem to gather additional information.
Gathering Information from the User

Information the user provides can be vital to your ability to correct a problem. Try to gather as much detail as possible on exactly what the problem is. Often when troubleshooting a problem, you might realize that what you’ve been troubleshooting for hours is not really the problem the user encountered. The more detail about the problem you can gather before you begin troubleshooting, the easier it is to find a resolution—and that means less frus-tration for you. Here is some general information to collect from users:

*

Details about exactly what the user experienced when the problem occurred.
*

Phone numbers for all parties involved in the problematic call or calls. You can use this as search criteria if you need to look through traces.
*

Actions performed by the user when the problem occurred. This includes what buttons were pressed and in what order.
*

User observations. This includes text messages displayed on the phone or recorded announcements.
*

Information about the user’s device. For example, if the user experienced a problem while using a 7960 phone, get the phone’s MAC address and IP address, along with registration information and any other statistics available from the phone.

Sometimes the information provided by an end user is not enough to even begin trouble-shooting. For example, if a user has trouble transferring calls, you should ask what steps the user took when the problem happened and, if possible, when the problem occurred so that you can examine traces. Sometimes the proper diagnostic tools are not enabled when the problem occurs, forcing you to ask the user to inform you the next time the problem occurs. Be sure to turn on tracing or debugs before making the request so that when the problem occurs again, you will have captured the data. Users can get quite irritated if you have to ask them for the same piece of information two or three times. Also point out to the user the importance of letting you know immediately after a problem occurs, as many of the diagnostic trace files overwrite themselves within several hours or days (depending on the amount of traffic on your system).
Determining the Problem’s Timeframe

In addition to what the problem is, you should try to determine when the problem occurred. Determining the problem’s earliest occurrence can help correlate the problem with other changes that might have been made to the system or other events that occurred around the same time. For example, assume that a regular workday begins at 9 a.m. and ends around 6 p.m. Many users report that they get a busy signal when dialing into their voice mail. It is important to know whether they are attempting to do this at 9:10 a.m., a time when the voice mail system is likely under attack from many users all trying to access the system at once. This might change the problem from a troubleshooting issue to a load-balancing or equipment-expansion issue. You check the voice mail system and notice that at the time the problem was reported, all the voice mail ports were in use. Clearly in this example you need more voice mail ports or servers to handle call volume. However, if the problem occurs at 10:30 p.m., capacity is likely not the problem, so it’s time to start troubleshooting your network and voice mail system. As another example, if a user reports that her phone was not working for 10 minutes and you know there was a network outage in her part of the building at that time, you can be relatively sure that the problem was due to the network outage.

When relying on end users to give “when” information about a problem, ask them to note the time on their phone when the problem occurred. The phone’s time is synchronized with the clock on the CallManager to which the phone is registered. As long as you have the time on your CallManagers and network devices synchronized, having a phone-based time from the user makes finding the proper trace files very easy.

In some cases, the information about when a problem occurred might be the only piece of information you have other than a limited description of the problem at hand. If you have information about when, you might be able to look through trace files during that timeframe to search for anything abnormal.

TIP

Although it is important to use information about when the problem started happening, it is equally important to not assume that the problem was a direct result of an event. For example, if a user reports a problem the day after an upgrade was performed on CallManager, you might give some credence to the notion that the upgrade might have caused the problem, but don’t automatically assume that this is the root cause.
Step 2: Analyzing the Data Collected About the Problem

Now that you have collected data from a variety of sources, you must analyze it to find the root cause and/or workaround for your problem.
Using Deductive Reasoning to Narrow the List of Possible Causes

The next part of your fact-finding mission is to identify the various components that might be involved and to eliminate as many components as possible. The more you can isolate the problem, the easier it is to find the root cause. For example, if a user complains about choppy voice quality, consider some of the following questions to help isolate the real problem, and think about how the answer will help narrow your focus:

*

Does the problem happen on only one phone? If so, you can probably eliminate hundreds or thousands of other phones as suspects. However, keep in mind a single user’s perspective. He might think the problem happens only on his phone, so you’ll have to ask other users to see if the problem is more widespread than a single phone.
*

What numbers are being called when the problem occurs? The answer to this question helps determine which parts of the system are being used when the problem occurs. For example, if the user never experiences poor audio quality when calling certain numbers but always experiences it when calling other numbers, this is a big clue.
*

Does the problem happen only between IP phones, only through one or more voice gateways, or both? The user probably won’t know the answer, but you’ll be able to answer this question yourself after you answer the preceding question about which numbers are being called when the problem happens.

You will find more detailed questions similar to these throughout this book when troubleshooting particular problems.

Although not all of the following apply to every problem, where applicable, you must check all of the following pieces involved in the call. Use your topology information to help obtain this information.

*

CallManager nodes involved in the signaling
*

Network devices that signaling and/or voice traffic traverse
*

Gateways or phones involved in the call
*

Other devices involved, such as conference bridges or transcoders

Concentrate your energy on the smallest subset of devices possible. For example, if all the users on a particular floor are having the same problem, concentrate on the problem a particular user is having. If you fix the problem for that one user, in most cases you fix it for all the affected users.
Verifying IP Network Integrity

One thing that people often forget is that your IP Telephony network is only as good as your IP network. A degraded network or a network outage can cause a wide range of problems, ranging from slight voice quality problems to a total inability to make or receive calls on one or more phones. The network is always a consideration when you encounter certain problems, so network health issues are covered throughout this book. Network health is especially important during the discussion of voice quality problems in Chapter 7, “Voice Quality,” because most voice quality problems stem from packet delay and/or loss.

Always remember to keep the IP network in mind and look at every layer in the OSI model, starting from Layer 1. Check your physical layer connectivity (cables, patch panels, fiber connectors, and so on). Then make sure you have Layer 2 connectivity by checking for errors on ports, ensuring that Layer 2 switches are functioning properly, and so forth. Continue working your way up the stack until you reach the application layer (Layer 7). As an example, two of the most common reasons for one-way audio (where one side of the conversation cannot hear the other) are the lack of an IP route from one phone to another and the lack of a default gateway being configured on a phone. Taking the layered approach, you would first check the cabling and switches to make sure that there are no errors on the ports. You would then check Layer 3, the network layer, by ensuring that IP routing is working correctly. When you reach this layer, you discover that for some reason the IP packets from one phone are unable to reach the other phone. Upon further investigation, you might discover that there was a missing IP route on one of the routers in the network or a missing default gateway on one of the end devices (such as an IP phone or voice gateway).
Determining the Proper Troubleshooting Tool

After you narrow down the appropriate component(s) causing a problem and have detailed information from the user(s) experiencing the problem, you must select the proper tool(s) to troubleshoot the problem. Most components have multiple troubleshooting tools avail-able to help you. Chapter 3, “Understanding the Troubleshooting Tools,” provides more details about some of the tools available for troubleshooting CallManager. You should use the tracing and debugging facilities available in CallManager and other devices to deter-mine exactly what is happening. Additional tools and traces are covered in the chapter associated with diagnosing certain types of problems. For example, Chapter 6, “Voice Gateways,” covers debugging Cisco IOS Software voice gateways. Because CallManager is central to almost all problems, information about various portions of the CCM trace facilities appears throughout this book.

This step is the most demanding on your troubleshooting skills because you analyze the detailed information provided in the various tools and use it to search for additional clues using other tools. Sometimes the problem description you have is not detailed enough to determine which tool to use. In this case, you should try various tools in search of anything that looks out of the ordinary.

The following case study shows how this troubleshooting methodology works in a real-world scenario.

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Deploying Cisco Voice over IP Solutions covers:

* Definitive guidelines on real-world VoIP deployments, the fundamentals of the latest VoIP solutions, and a look into the future of VoIP services
* Different techniques for engineering and properly sizing traffic-sensitive voice networks
* Basic concepts applicable to echo analysis, echo cancellation, and locating and eliminating echoes
* Various QoS features applicable to voice
* Detailed information on call admission control (CAC)
* Dial plan configuration recommendations on Cisco H.323 gateways and gatekeepers used to support large dial plans
* Basic tasks of designing a long-distance VoIP network
* The two classes of hosted voice networks: Managed Multiservice (MMS) networks and packet voice VPNs
* Fax services store and forward as well as real-time relay fax services
* Sample configurations and step-by-step examples to help you learn how to build a VoIP network

Deploying Cisco Voice over IP Solutions provides networking professionals the knowledge, advice, and insight necessary to design and deploy voice over IP (VoIP) networks that meet customers’ needs for scalability, services, and security. Beginning with an introduction to the important preliminary design elements that need to be considered before implementing VoIP, Deploying Cisco Voice over IP Solutions also demonstrates the basic tasks involved in designing an effective service provider-based VoIP network. You’ll conclude with design and implementation guidelines for some of the more popular and widely requested VoIP services, such as prepaid services, fax services, and virtual private networks (VPNs).

This book is a collaboration of Cisco Systems CCIE(r) engineers, technical marketing engineers, and systems engineers. You’ll find design experience from people who have designed some of the world’s largest VoIP networks.

Networks, whether voice or data, are designed around many different variables. Two of the
most important factors that you need to consider in network design are service and cost.
Service is essential for maintaining customer satisfaction. Cost is always a factor in
maintaining pro?tability. One way you can maintain quality service and rein in cost in
network design is to optimize circuit utilization.
This chapter describes the different techniques you can use to engineer and properly size
traf?c-sensitive voice networks. You’ll see several different traf?c models and explanations
of how to use traf?c probability tables to help you engineer robust and ef?cient voice
networks.

Traf?c Load Measurement
In traf?c theory, you measure traf?c load. Traf?c load is de?ned as the ratio of call arrivals
in a speci?ed period of time to the average amount of time it takes to service each call
during that period. These measurement units are based on Average Hold Time (AHT). AHT
is de?ned as the total amount of time of all calls in a speci?ed period divided by the number
of calls in that period. For example:
3976 total call seconds / 23 calls = 172.87 sec per call = AHT of 172.87 seconds
The two main measurement units used today to measure traf?c load are the following:
• Erlangs
• Centum Call Seconds (CCS)
In 1918, A.K. Erlang developed formulas that could be used to make predictions about
randomly arriving telephone traf?c. The Erlang—a measurement of telephone traf?c—was
named in honor of him. One Erlang is de?ned as 3600 seconds of calls on the same circuit,
or enough traf?c load to keep one circuit busy for 1 hour.
Traf?c in Erlangs = (number of calls × AHT) / 3600
Example: (23 calls × 172.87 AHT) / 3600 = 1.104 Erlangs
CCS is based on 100 seconds of calls on the same circuit. Voice switches generally measure
the amount of traf?c in CCS.
Traf?c in CCS = (number of calls × AHT) / 100
Example: (23 calls × 172.87 AHT) / 100 = 39.76 CCS
Which unit you use depends on the equipment you use and the unit of measurement it
records in. Many switches use CCS because it is easier to work with increments of 100
rather than 3600. Both units are recognized standards in the ?eld. The following is how the
two relate:
1 Erlang = 36 CCS
Although you can take the total call seconds in an hour and divide that amount by 3600
seconds to determine traf?c in Erlangs, you can also use averages of various time periods.
These averages allow you to utilize more sample periods and determine the proper traf?c.

More info?Deploying Cisco Voice over IP Solutions (Networking Technology)
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* Includes step-by-step configuration instructions for CallManager features and Unity administration tasks
* Demonstrates how to deploy devices and implement your dial plan
* Covers Call Admission Control features and class of service
* Examines different subscriber types and how subscribers are added, imported, and managed
* Includes step-by-step instructions for call handling and auto attendant configuration
* Describes how to use Unity and CallManager together to deliver unique features

To properly deploy any type of technology, networking professionals must understand not only the technology but also how to configure and integrate it with other solutions. That’s the key to Configuring CallManager and Unity–it focuses on the configuration issues associated with CallManager and Unity® deployments while ensuring that you understand the technologies behind your deployment.

Configuring CallManager and Unity includes step-by-step guides that system administrators and other networking professionals can use in the field. These step-by-step instructions have been worked out by an author who has both taught and implemented Cisco® solutions in real-world situations, so coverage is comprehensive for both basic and complex implementations. You will find information that will assist in the configuration of CallManager-related tasks, such as device configuration, gateway implementation, and dial-plan creation to name a few. You will also find Unity-related configuration tasks, ranging from the basics, such as holiday and schedule configuration, to more involved tasks, such as Simple Mail Transfer Protocol (SMTP) networking implementation. In addition to covering Unity and CallManager tasks, this book includes a chapter on leveraging the capabilities of both systems to create integrated solutions, such as a MeetMe conference manager.

Tasks in Configuring CallManager and Unity are organized in the same order you would naturally perform them, and some tasks are cross-referenced with other required tasks for easy reference. You learn not only how to configure CallManager and Unity but also how to create a more feature-rich environment by leveraging CallManager and Unity features. Regardless of your specific needs, you’ll find Configuring CallManager and Unity to be a timesaving tool when performing common or complicated configuration tasks.

This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press help networking professionals understand voice and IP telephony technologies, plan and design converged networks, and implement network solutions for increased productivity.

After all predeployment tasks are completed, devices can be added to the system. This chapter discusses the tasks required to add certain devices to the CallManager cluster.

A number of types of devices can be added to the CallManager. The devices discussed in this chapter fall into one of two categories, clients or gateways. Other devices, such as gatekeepers, are discussed in future chapters.

We start by looking at clients, more specifically phones. There are a number of different models of Cisco IP phones, but the task of adding each is very similar. Next, gateways are covered. Gateways allow connectivity to another system such as the Public Switch Telephone Network (PSTN) or another private branch exchange (PBX). In CallManager 4.0 there are over 30 different types of gateways, and although the function of each is similar, the configuration of each varies. It is not possible to provide a step-by-step guide for the configuration of each, but this chapter does include detailed steps on how to configure the most popular types.

In all the steps in this book, I have tried to cover all the parameters that appear on the screen. Because each version of CallManager adds fields, all the fields you see in these steps may not appear on your screen, or additional fields may be present, due to the version of CallManager you are running. Regardless of the version you are running, the steps in this chapter will help you walk through the process. Not all the parameters must be configured. Some can remain at default, whereas others are not required at all. The end of the names of required parameters are marked with an asterisk (*). In some cases you may need to configure only the required parameters for a device to function. It is a good idea, however, to review each parameter so that you can be certain the device is configured exactly the way you want it.
Adding Clients

Depending on the environment, adding a phone can be as simple as plugging it into a port that has connectivity to the CallManager. Although it is possible to configure a CallManager to allow phones to be automatically added simply by connecting them to the network, it is not always wise or desired. This section explores four ways phones can be added to the system, but before adding phones to a system, there are a few more components to configure.

You are probably thinking, “Wait a minute, the last chapter discussed predeployment tasks.” That’s right, it did, but those were the general or global settings that should be configured. This section looks at settings that are specific to phones.

Which came first the chicken or the egg? This is a question that has plagued the minds of great thinkers from the beginning of time. Okay, maybe it hasn t, but a similar issue sometimes occurs when teaching new technologies. When two separate concepts are interdependent, it may be difficult to grasp either concept until you understand both of them. This being said, you will notice throughout this chapter that new concepts and components are mentioned that are not discussed in detail until later in this book. In these cases, a brief description is offered for each new concept, as is a reference to the chapter that lends greater detail. You may choose to jump ahead to gain a better understanding or just accept the brief description, knowing more detail is offered later.
Defining Device Settings

Before adding phones, it is recommended, but not required, that some device settings be configured. Configuring the device setting first will most likely save time. These settings include Phone Button Templates, Softkey Templates, and Device Defaults. The following section explains the function of each of these and how to configure them.
Phone Button Templates

Some Cisco IP phones have buttons that can be configured for various functions. The most popular functions these buttons support are lines and speed dials. When configured as lines, extension numbers can be assigned to the buttons. Each phone must have at least one button configured as a line. When a button is configured as speed dial, administrators or users can assign a speed dial number to it, and because most Cisco phones have only a few of these buttons (one to eight), the number of speed dials that can be defined in this way is limited. The user can access additional speed dials by using a feature known as abbreviated dial, which is discussed later in this chapter.

In addition to lines and speed dial functions, these buttons may be able to be configured for other functions, depending on the phone model. There are also phones on which the buttons are not configurable. Table 3-1 shows which phones have configurable buttons and which do not.
Table 3-1. Configurable/Non-Configurable Phone Buttons

Configurable Button Phones

Fixed Button Phones

30 VIP

7902

12 SP(+)

7905

7910

7912

7920

7935

7940

7936

7960

ATA 186-188

7970

VCG

The exact function that can be assigned to buttons varies depending on the model of the phone. The 7940, 7960, and 7970 buttons can be configured as Service URL, privacy, speed dial, or line buttons. By configuring a button as a Service URL, a user may access a phone service by simply pressing the button. A button that is configured as a privacy button allows a user to make a call private. This feature is used on shared lines and prevents the other shared-line phone from entering the call. It is also possible that a button has no function assigned to it. When a button serves no function, it is labeled as “none.”

The 7920 buttons can be configured only as lines or speed dials.

When a phone is added to the system, a phone button template is associated to the phone. The template is used to determine the function each button will serve. If the phone is added via auto-registration, it uses the template defined under device defaults. Device defaults settings are discussed later in this chapter. The creation of these templates is simple because, on most phones, there are only two functions for these buttons.

To create phone button templates follow these steps:

Step 1. From within CCMAdmin, select Device>Device Settings>Phone Button Template.

Step 2. Select the Add a New Phone Button Template link.

Step 3. On the next page you must select an existing phone button template to copy. This is because a new phone button template must be based on an existing template. From the drop-down list in the field labeled Phone Button Template, select the standard template for the correct model of phone. For example, if you are creating a phone button template for a 7960, select the Standard 7960 Template.

Step 4. Then click the Copy button.

Step 5. A screen like that shown in Figure 3-1 displays. At first glance you may think that you have selected the wrong phone type because it shows 34 buttons instead of the six you were expecting. Because the 7960 can support two 7914 expansion modules, which support 14 buttons apiece, 34 total buttons are possible. In the Button Template Name field enter a descriptive name.

03fig01.jpg

Figure 3-1 Phone Button Template Configuration

It is recommended that the Button Template Name identify the configuration of this template. For example, if the template is configured with two lines, three speed dials, and a privacy button, a good name would be “7960 2-3 w/Privacy.”
Step 6. Assign the proper function to each line by selecting Service URL, Privacy, Line or Speed Dial, or None from the drop-down list next to the button number.

Step 7. Next to the feature field is a label field. Enter a descriptive label in this field such as Speed Dial 2 or Line 3.

Step 8. Click the Insert button to save the new template.

Softkey Template

In addition to the buttons that can be used for lines and speed dials, Cisco IP phones have buttons that are referred to as softkeys. These buttons (keys) allow the user to access features of the phone such as hold, transfer, conference as well as many others. The function of each key changes depending on the state of the call. Because there are often more available functions configured than physical softkeys, the last softkey functions as a toggle key allowing the user to scroll through all the available options. The softkey template allows you to determine what functions are available on the phone and in what order they display. Much like phone button templates, the softkey templates can be associated directly with a phone. However, as you saw in the last chapter, the softkeys templates are also associated to device pools, which are in turn associated with phones. By associating the softkey template to a device pool, you can easily and quickly assign the softkey template to a large number of phones. If different softkey templates are associated with a phone at both the phone level and the device pool, the one at the phone level takes precedence.

The following steps take you through the process of creating a softkey template.

Step 1. From within CCMAdmin, select Device>Device Settings>Softkey Template.

Step 2. Click the Add a New Softkey Template link.

Step 3. On the next page you must select an existing softkey template on which to base the new template. Select a template from the drop-down list in the Create a softkey template based on field.

Step 4. Then click the Copy button.

Step 5. On the next screen, enter name and description for the new template. The description should help identify the features associated with the template.

The Application field lists the applications that are available in this template and cannot be changed in this screen.

Step 6. Click the Insert button to add the new template.

Step 7. After the template is added, you must configure it. If you want to add applications that are found on other softkey templates, click the Add Application button. If you do not wish to add applications from an existing template, skip this and the next step. For example, you would not add an application if you are simply moving or adding standard softkeys to the template.

Step 8. If you selected Add Application in the new window that displays, select the standard softkey template that contains the application you want to add to the new template and click Insert and Close.

Step 9. Click the Configure Softkey Layout link to modify the current layout.

A screen similar to that found in Figure 3-2 displays. On the left side of the screen, all of the states of the phone are listed. Because the softkeys change depending on the state of the phone, each state must be configured separately. In Figure 3-2 the new template allows access to the call-back feature. In this example the call-back softkey must be configured for all states from which it might be accessed which are On Hook and Ring Out.

03fig02.jpg

Figure 3-2 Softkey Layout Configuration
Step 10. Determine which call state you want to modify and select it from the list. For instance, if you wanted to add a softkey that would appear on the phone while you were on a call, you would select Connected.

Step 11. The softkey currently assigned displays in the box on the right side of the screen labeled Selected Softkey. The softkeys that may be added display in the box on the left side labeled Unselected Softkeys. To add unselected softkeys, click once on the softkey and then click on the top arrow of the two arrows that display between the two boxes.

Because there are a limited number of softkeys on the phone, it is common to have more softkeys assigned than buttons on the phone. In this case the last button becomes a “more” button that acts as a toggling mechanism to allow users to access the other features. It is important to understand that the more button will automatically appear on the phone and is not something you configure on the template.
Step 12. To remove a current softkey, click on the softkey you wish to remove and click the bottom arrow of the two arrows that display between the boxes.

Step 13. The softkeys display on the phone in the same order that they display in the Selected Softkeys box. To change the order in which they display, highlight the desired softkey and click the up or down arrow that displays on the left side of the Selected Softkey Box.

If softkeys are used in more than one call state, for instance the callback softkey is available in the On Hook and Ring Out call states, pay attention to the position of the softkey in each of the call states. It’s a good idea to keep the softkey in the same position for each call state if possible. If it is desirable to leave a softkey position unused, the Undefined softkey can be used as a placeholder.
Step 14. After you have modified all the desired call states, click the Update button.

When modifying an existing softkey template, you need to reset the devices that are associated with the templates. To do this, click the Restart Devices button on this page. Take care when resetting devices, because resetting causes the phones to be unusable while they reset. This will not affect phones that are currently on a call. The reset will occur once the phone is idle. It is always recommended that whenever a reset must be performed on a large number of phones, it be done off hours to minimize the impact on the end user.
Device Defaults

When a phone boots, it requests a configuration file from the CallManager Trivial File Transfer Protocol (TFTP) server. The file that it requests is associated with the phone’s Media Access Control (MAC) address. If the phone has not previously been registered with a CallManager on the system and has not been manually added, no configuration file exists. When auto-registration is being used, a phone uses a default configuration file that defines how it is to attach to the system and register. The Device Defaults settings define the values used.

Three settings are defined under the Device Defaults settings as follows:
Load Information

This is the ID of the firmware load that the device should be running. When CallManager is shipped, it includes a current version of the firmware loaded for each device. From time-to-time the firmware is upgraded to offer additional features or patches. By editing this field you can specify the ID of the new load. The next time the device boots, it downloads the new firmware load.
Device Pool

This field allows you to define which device pool is used when auto-registration takes place.
Phone Template

This field allows you to define what phone button template devices use when auto-registration takes place.

The following steps show how the device defaults are configured.

Step 1. From within CCMAdmin, select System>Device Defaults.

Step 2. A screen similar to that shown in Figure 3-3 displays. Most often the Load Information can remain at default. However, from time-to-time new loads need to be deployed. In this case, locate the name of the device for which you are updating the load and enter the new load ID in the Load Information field. Make sure you enter the ID correctly and that the ID is for the correct device. Entering an invalid or incorrect ID in this field can cause the device to fail.

03fig03.jpg

Figure 3-3 Device Defaults Configuration

When a new load is to be used, it must first be copied to the TFTP server so that the device will be able to download it.
Step 3. Next, you edit the Device Pool. To change the default device pool for a device, simply click the down arrow in the Device Pool field associated with the device and select the desired device pool from the list that displays.

Step 4. The last item that can be selected on this page is the phone button template. To change this item for a device, click the down arrow in the Phone Template field associated with the device and select a template from the list that displays.

Adding Phones

When a phone is added to the system, information about the phone is entered into the SQL database. This information defines nearly every aspect of the device. Phones can be added to the system in a number of ways, but the net effect is the same.

Four methods for adding phones are explored in this chapter starting with the method known as auto-registration. Before delving too deeply into this method, let’s take a quick look at all four.

1. Auto-Registration— allows phones to be plugged into the system and automatically register. An extension is assigned to the phone from a range defined by the administrator.
2. Manual— all information for the phones manually entered before the phone is plugged in.
3. Bulk Administration Tool (BAT)— information for a large number of phones inserted using BAT. This is done by entering the information into a Comma Separated Value (CSV) file and then using BAT to insert the information into the CallManager database.
4. Tools for Auto-Registration Phone Support (TAPS)— Similar to BAT except that the MAC address is not entered in the CSV file and additional steps are required when the phone is plugged into the network. However, even with the additional steps, it is quicker than just using BAT when performing a large installation.

Auto-Registration

As the name implies, using this method allows phones to register to the system by merely being plugged into the CallManager network. Although this method has the advantage of allowing you to quickly add phones, it has its disadvantages as well. When a phone is added this way, it is assigned an extension number from a range of numbers that you define. It assigns these extensions in a first-come-first-served fashion. For instance, if you defined the range to be from 2000 to 2100, the first phone to register would receive the extension 2000, the second 2001, and so on. You can see how this may not be desirable. If you are fortunate enough to be performing a deployment in which you can assign extensions as you wish, then this may not be a concern. However, in situations such as when you are replacing an existing PBX, you will most likely want to reuse the existing extensions, hence the auto assignment of extensions may not be desirable. Auto-registration may still be used in these environments; you will just need to modify the extensions that were automatically assigned.

Rogue phones are another anomaly that can arise when auto-registration is used. If improperly configured, it is possible for a phone to be added to the system without your knowledge. The addition would, of course, have to be made by someone who has a Cisco IP phone and is able to plug it into your network. Although this may seem less than likely, it is still important to ensure that the dialing capabilities be restricted for any rogue phones that may find their way onto your system. This is done by defining a Calling Search Space (CSS) for auto-registration in the device pool. CSS defines a device’s calling privileges. This concept is discussed in greater detail in Chapter 5: Configuring Class of Service and Call Admission Control. Choose a CSS that has limited access, perhaps only internal or local access. This, however, also has a drawback. Limiting the CSS of auto-registered phones means that all phones that are added using auto-registration will have a limited CSS. This means for a phone that was added through auto-registration to have greater calling privileges, the CSS must be changed. Often auto-registration is used during the initial deployment and then turned off.

As you can see, using auto-registration has both benefits and drawbacks. The choice of whether to use it depends on the environment and is ultimately up to you.

If all the previously discussed predeployment tasks have been completed, very little additional configuration is required to implement auto-registration. All that needs to be done is to assign an extension range and enable auto-registration at the CallManager Group. If you choose to use auto-registration, the steps that follow help walk you through these tasks.

Step 1. From within CCMAdmin, select System>Cisco CallManager.

Step 2. Enter search criteria in the search field, and click Find. You may also leave the search field blank and click Find. This results in all CallManagers being displayed.

Step 3. From the list that displays, select the CallManager on which you want to enable auto-registration.

Step 4. A screen similar to that shown in Figure 3-4 displays. Enter the starting extension number in the Starting Directory Number field and the ending extension in the Ending Directory Number field.

03fig04.jpg

Figure 3-4 CallManager Configuration for Auto-Registration
Step 5. You should notice that in the Auto-Registration Information field the Auto-Registration Disabled checkbox is unchecked at this point. This checkbox automatically becomes unchecked when you enter a starting and ending extension number. DO NOT check this box. Checking this box resets the starting and ending extension numbers.

Step 6. Click the Update button to save settings.

In the next section, you will be selecting a CallManager Group for auto-registration. If the CallManager Group that you select contains more than one CallManager, you should enter the starting and ending extension numbers for each of the CallManagers in the group.

After an extension range is defined, you need to enable auto-registration for the CallManager group. The steps that follow show how this is done.

Step 1. From within CCMAdmin, select System>Cisco CallManager Group.

Step 2. Enter search criteria in the search field and click Find. You may also leave the search field blank and click Find. This results in displaying all CallManager Groups.

Step 3. From the list that displays, select the CallManager Group on which you want to enable auto-registration.

Step 4. A screen similar to that shown in Figure 3-5 displays. Check the Auto-registration Cisco CallManager Group box.

03fig05.jpg

Figure 3-5 CallManager Group Configuration for Auto-registration

Only one CallManager Group may be selected as the Auto-registration CallManager Group. If you check this box and there is already an existing Auto-registration CallManager Group, a window displays stating, “You have selected this Cisco CallManager Group to be the Auto-registration Cisco CallManager Group. The old Auto-registration Cisco CallManager Group will be deselected.” Only check that box for the CallManager Group that is responsible for auto-registration.
Step 5. Click the Update button to save changes.

Manually Adding Phones

In addition to using auto-registration to add phones, you can manually add them. You often add phones manually for a small number of phones (such as 20). If all the predeployment tasks have been completed, the manual adding of phones is quite simple. This section takes you through the steps of the process of manually adding phones.

Manually adding phones has very few drawbacks, aside from the fact that it can be somewhat time consuming. You must enter the MAC address and other information, such as device pool and directory numbers, when you add each phone. Because CCMAdmin is a web-base interface, you must wait for the page to reload when you update information. Although a second or two may not seem too long, when you have to wait for multiple page reloads, it can start to add up. That is just the nature of this type of interface and little can be done about it.

The steps that follow take you through the process of manually adding a phone and explore each field that can be populated for each phone. A brief description is included for any field that has not yet been discussed in this book. The phone used in these steps is a 7960, but the process is very similar for most of the more popular Cisco IP Phone models. When you add other phone models, a field may not display. This simply means that the model does not require or support that particular field. Because there are a number of steps to this process, section headings are used to mark the point at which each new set of parameters begins. These same headings display on the configuration screen as well, which should help you keep track of where you are.

Step 1. From within CCMAdmin, select Device>Add a New Device.

Step 2. Select Phone from the drop-down list and click Next.

Step 3. A new page displays. From the drop-down list, select the type of phone you want to add. Click Next.

Device Information

Step 4. A screen similar to that shown in Figure 3-6 should display. The first field that must be entered is the MAC Address of the phone. The MAC address can be found on the back of the phone as well as the box in which the phone was packaged.

03fig06.jpg

Figure 3-6 Phone Configuration

It seems that over the years the numbers on the back of the phone have been getting smaller. Then again it seems everything I am reading nowadays is getting smaller, so maybe it’s not the phone. If you find the same to be true for you, you can also get the MAC address to display on the LCD of the phone. The method for this differs among the various models. On the 7960 and 7940, press the settings and then the 3 on the keypad. Of course, power must first be supplied to the phone.
Step 5. The next field is the Description field. Enter a description that will help you quickly identify the phone later.

If you do not enter anything in the Description field, a default description that is the MAC address preceded by SEP is entered automatically. SEP stands for Selsius Ethernet Phone. Selsius was the name of the company that made CallManager before Cisco bought the company and the product. A description I like to use follows the following format: Last Name, First Name Extension Number (for example, Smith, John 2012). This format allows me to search by name and easily see the extension number from the search page.
Step 6. In the Owner User ID field, enter the user ID of the primary user of the phone. The user must already exist in the directory that is used with CallManager, which by default is DC directories. Creating users is covered in Chapter 6: Configuring CallManager Features and Services. An easy way to assign a user is to click the Select User ID link to the left of the field and then search for the desired user. This information is included in Call Detail Records (CDRs). Leave this field blank if extension mobility is going to be used.

Step 7. From the Device Pool drop-down list, select the device pool the phone will use.

The device pool, MAC address, and phone button template are the only required fields, aside from product-specific information that can be found at the bottom of the configuration screen. Because the product-specific information is automatically set to default values, you could add the phone at this point without configuring any other parameters. This is useful if you need to quickly add phones and you are certain that the default system values, and those set in the device pool, are adequate for this phone.
Step 8. The CSS determines the destinations that can be dialed from the phone. CSS are discussed in Chapter 5: Configuring Class of Service and Call Admission Control. Choose a CSS from the Calling Search Space drop-down list. If this field is left at None, the dial privileges of this phone could be limited.

Step 9. The Automated Alternate Routing (AAR) is used to provide an alternate route if a call fails due to insufficient bandwidth. The AAR CSS can be used to limit the paths a call may use when it is rerouted. ARR is covered in Chapter 6: Configuring CallManager Features and Services. Select an AAR CSS from the AAR Calling Search Space drop-down list.

Step 10. The next field is Media Resource Group List. This determines to which media resources this phone will have access. Media resources are discussed in further detail in Chapter 5: Configuring Class of Service and Call Admission Control. From the Media Resource Group List drop-down list, select the desired group. If no media resource group list is chosen, the list defined in the device pool is used.

Step 11. The next two fields allow you to configure what audio source is heard when a call is placed on hold. The first of the two, which is labeled User Hold Audio Source, determines what is heard when the call is placed on hold by pressing the hold button. The second of the two, Network Hold Audio Source, determines what audio is heard when the call is placed on hold by pressing the transfer, call park, or conference button. Select the desired audio source from the drop-down list for each field. If no audio source is chosen, the source defined in the enterprise parameters is used.

Step 12. Information entered in the Location field is used to prevent WAN links from becoming oversubscribed in centralized deployments. These locations are discussed more in Chapter 5, “Configuring Class of Service and Call Admission Control.” If you have defined locations, select the appropriate one for this phone from the drop-down list.

Step 13. The User Locale field determines the language and fonts used for the phone. The default user locale, which is set in the enterprise parameters, is used if this field is left set to None. If this phone needs to use a different locale than is defined by its device pools, or the enterprise parameters, select the proper one from the drop-down list.

Step 14. The Network Locale field determines what locale is used for this phone. This impacts the tones and cadences used. The default network locale was defined in the enterprise parameters. If a different value is selected here this value takes precedence. If this field is set to None, the enterprise parameter setting is used. If this phone needs to use a different locale than is defined by its device pools, or the enterprise parameters, select the proper locale from the drop-down list.

Step 15. The next field labeled Device Security Mode is used to determine if any security is used for calls placed from this phone and, if so, what type. Select the desired security mode or leave it set as Use System defaults.

Step 16. The next field is the Signal Packet Capture Mode. This field is for trouble shooting purposes only and should not be configured when adding a new phone.

Step 17. The Packet capture duration field is for trouble shooting purposes only and should not be configured when adding a new phone.

Step 18. The Built-In Bridge field is used to enable and disable the built-in bridge. This bridge can be used when the barge feature is invoked. Select the desired state of the built-in bridge from the drop-down list.

Barge is a feature that allows a phone to join an active call on another phone if the two phones have a shared line.
Step 19. If the Retry Video Call as Audio checkbox is checked, a video call will try to connect as an audio call if it cannot connect as a video call.

Step 20. If the Ignore Presentation Indicators (internal calls only) check box is checked, internal caller ID restrictions are ignored. This means that if an internal call is configured to block caller ID, the caller ID will still show up on this device.

Step 21. The Privacy field is used to determine if the phone can enable privacy for calls on a shared line. Select the desired state for this field from the drop-down list.

Phone Button Template Information

Step 22. In the next field, a phone button template is selected for this phone. From the drop-down list in the Phone Button Template field, select the desired template.

Not all the templates will display in this field, only those will that apply to the model of phone you are adding.

Softkey Template Information

Step 23. From the drop-down list in the Softkey Template field, select a softkey template for this phone.

Expansion Module Information

Step 24. The next two fields labeled Module 1 and Module 2 are used when expansion modules (7914) are used with the phone being added. If the phone has expansion modules, select the modules from the drop-down list.

Firmware Load Information

Step 25. The next three fields are used to define which firmware load ID the phone and its expansion modules use. In most cases, these fields should be left blank. When left blank, the Load ID specified on the Device Defaults Configuration page is used for this phone. If the need ever arises to set a specific phone to a specific load ID, the load ID should be entered in this field.

Cisco IP Phone—External Data Locations

Step 26. The next set of fields is used to define data locations for the phone. This information is used to determine where the phone should search for certain data, such as help screens and phone services. In most cases these fields can be left blank and the system defaults will be used. Table 3-2 lists these fields and a brief description of each. If values other than the system defaults need to be used by this phone, enter them in the appropriate fields. Any values entered in these fields will be used for this device and will override the values found in the enterprise parameters page.

Table 3-2. Cisco IP Phone—External Data Locations Parameters

Parameter

Description

Information

The URL the phone uses when the (i) is pressed on the phone

Directory

The location of the directory the phone uses

Messages

The URL that is used when the Messages button is pressed. Because you normally want a number dialed when this button is pressed, this field should be left blank.

Services

The URL where services can be found

Authentication Server

URL of the authentication server for requests made to the phone web server

Proxy Server

The proxy server used by the phone

Idle

The URL that is displayed on the phone after the Idle Timer expires

Idle Timer

The amount of time in seconds that the phone must remain idle before the Idle URL is displayed

Certificate Authority Proxy Function (CAPF) Information

Step 27. The next set of parameters deals with Certificate Authority Proxy Function (CAPF). These settings are used to configure certificate specific information. Certificates are used to help prevent the tampering of call signaling and media streams. For more information on CAPF refer to the “Cisco IP Phone Authentication and Encryption for Cisco CallManager” guide at Cisco.com. This guide can be found by searching Cisco IP Phone Authentication and Encryption at Cisco.com. If CAPF is not being used, these fields do not need to be configured. The Certificate Operation field is used to install, upgrade, or delete a certificate. The available options are:

No Pending Action— Displays when no certificate operation is currently active.

Install/Upgrade— Select this when you want to install or upgrade a certificate.

Delete— When selected the current certificate will be deleted.

Troubleshoot— Allows certificate information to be viewed in a CAPF trace file.

Step 28. The Authentication Mode field determines the method that will be used by the phone to authenticate with CAPF. Choose one of the following:

- By Authentication String
- By Null String
- By Existing Certificate (Precedence to LSC)
- By Existing Certificate (Precedence to MIC)

Step 29. The Authentication String is used only when the By Authentication String is chosen in the previous step. Enter the string you wish to use. It must be between 4 to 10 digits.

Step 30. The Key String field determines the key size for the certificate. The valid choices are 512, 1024 and 2048. Select the desired value.

Step 31. The Operation Completes By field specifies when the install, upgrade, or delete must be complete. Enter the desired date and time in this field.

Step 32. The Certificate Operation Status field displays the progress of the certificate operation. Nothing is entered in this field; it is read only.

Multilevel Precedence and Preemption Information

Step 33. The next three fields define Multilevel Precedence and Preemption (MLPP) characteristics of the phone. If these fields are left blank or set to default, the values set in the device pool are used. If MLPP is not being used, these fields may be left blank. The first MLPP field is the MLPP Domain. MLPP grants only higher priority from calls within the same MLPP domain. For this reason a MLPP domain is needed.

Step 34. The second field in this category, which is called MLPP Indication, determines whether tones and indications are presented when a precedence call is made. The precedence indication may be a special ring back or a display if the caller’s phone supports it, and a special ringer on the called parties side.

Step 35. The third MLPP field is MLPP Preemption. This parameter determines whether a higher precedence call preempts a lower precedence call. The value of Disabled does not allow this to happen. To cause a lower precedence call to be terminated if a higher precedence call requires the resources, set this parameter to Forceful.

Product Specific Information

Step 36. The last set of fields labeled Product Specific Configuration are specific to the model of phone you are configuring. To see an explanation of each of these fields click the [i] icon located to the right of the category title.

Step 37. After all settings have been defined, click the Insert button at the top of the screen. A message displays informing you that the phone has been added and asking you if you would like to add extensions. Press OK and continue using the steps outlined in the following sections to add and configure a line on a phone.

Add a Line to a Phone

After a phone is added, a line must be configured for it. The following steps show how to add a line to a phone. Because there are a number of steps to this process, section headings are used to mark the point at which a new set of parameters begins. These same headings also display on the configuration screen, which should help you keep track of where you are.

Step 1. If you are adding a new phone and have used the steps in the preceding section, you should see a screen similar to that shown in Figure 3-7. To add a line to an existing phone, follow steps 2 though 4 to reach this screen. If you are already at this screen, skip to step 5.

03fig07.jpg

Figure 3-7 Directory Number Configuration
Step 2. From within CCMAdmin, select Device>Phone.

Step 3. Enter search criteria in the search field to limit the results, and click the Find button.

Step 4. From the list that is generated, select the phone to which you want to add a line. On the left side of the Phone Configuration screen the available lines are listed. Choose a line that has the label Add new DN.

If all lines have an extension number already assigned, no more lines can be added to this device. If the phone button template assigned to this phone has some of the buttons defined as speed dial, you may be able to add more lines by changing the phone button template.
Directory Number

Step 5. The first and only required field is the Directory Number. Enter the extension number in this field.

Step 6. The Partition field defines the partition to which this directory number is assigned. The partition is used to determine what devices may call this extension. Partitions are discussed in more detail in Chapter 5: Configuring Class of Service and Call Admission Control.

Directory Number Settings

Step 7. The Voice Mail Profile field determines which voice mail profile the directory number uses. The voice mail profile defines the number that is dialed when the messages button on the phone is pressed. Voice mail profiles are discussed in further detail in Chapter 5: Configuring Class of Service and Call Admission Control. Select the voice mail profile from the drop-down list.

Step 8. The next field allows a CSS to be assigned as the line level. This determines what destinations can be reached when calling from this line. Select the Calling Search Space from the drop-down list.

It is important to understand what happens when a CSS is assigned to the line and the device. In short, the two CSSs are combined, however there is a little more to it than this. For a detailed explanation, refer to Chapter 5: Configuring Class of Service and Call Admission Control.
Step 9. The AAR Group field determines the AAR group with which the line is associated. An AAR group defines the prefix that is assigned when a call fails due to insufficient bandwidth. AAR is discussed in further detail in Chapter 6: Configuring CallManager Features and Services. Select an AAR group if AAR is being used. If this field is set to None, AAR is, in effect, disabled on the line.

Step 10. The next two fields allow you to configure what audio source is heard when a call is placed on hold. The first of the two, which is labeled User Hold Audio Source, determines what is heard when the call is placed on hold by pressing the hold button. The second of the two, Network Hold Audio Source, determines what audio is heard when the call is placed on hold by pressing the transfer, call park, or conference button. Select the desired audio source from the drop-down list for each field. If no audio source is chosen, the source defined at the device level is used, and if None is chosen there, the source set in the device pool is used.

Step 11. The Auto Answer field determines if the line automatically answers incoming calls without the handset being lifted. This parameter can be set to auto answer using the speakerphone or headset. If you want the line to auto answer, select either Auto Answer with Headset or Auto Answer with Speakerphone from the drop-down list. If auto answer is not desired on the line, leave this field set to Auto Answer Off.

Call Forward and Pickup Settings

Step 12. The next seven fields deal with call forwarding. These fields determine the forwarding destination, which depends on the reason for the forward. The seven types of forwards are:

Forward All— forwards all incoming calls

Forward Busy Internal— forwards calls from internal callers when the line is busy

Forward Busy External— forwards calls from external callers when the line is busy

Forward No Answer Internal— forwards calls from internal callers that are not answered

Forward No Answer External— forwards calls from external callers that are not answered

Forward No Coverage Internal— forwards calls from internal callers when a CTI route point has no coverage

Forward No Coverage External— forwards calls from external callers when a CTI route point has no coverage

The no coverage forwards are needed only when configuring a line for a computer telephony integration (CTI) route point.
You may configure each type of “forward” to forward calls to voicemail or a specific extension. To forward to voice mail, check the Voice Mail box. For this to work, a voice mail profile must be defined for the line. To forward calls to another extension, enter the extension number in the Destination field. When a destination is entered into any of the internal forwards, the number is automatically entered into the corresponding external forward. If you wish the external calls to be forwarded to a different destination, simply enter the desired destination in the appropriate external forward field. A calling search space can be applied to each forward type, which limits the destinations to which a call can be forwarded. This is useful when you want to restrict a line from forwarding calls to numbers that are long distance, but still want long distance calls to be placed from the line.

Step 13. Enter the appropriate destinations and calling search spaces for each forward type.

Step 14. In the No Answer Ring Duration field, enter the number of seconds that the line will ring before forwarding to the Forward No Answer destination. If this field is left blank, the value configured in CallManager service parameter is used.

Step 15. The Call Pickup Group field determines which call pickup group this directory number belongs to. Call pickup groups allow a user to redirect an incoming call on another phone to the user’s phone. Select the desired call pickup group from the drop-down list. Call pickup groups are covered in more detail in Chapter 6: Configuring CallManager Features and Services.

MLPP Alternate Party Settings

Step 16. The next set of parameters deals with MLPP alternate party settings. These settings allow you to configure an alternate destination for precedence calls that are not answered on this line or the forwarded number assigned to this line. If MLPP is not being used, these parameters can be left empty. In the first field, which is labeled Target (Destination), enter the number to which unanswered precedence calls should be forwarded.

Step 17. In the MLPP alternate party Calling Search Space field, select the appropriate search space from the drop-down list. This calling search space limits the destinations to which precedence calls can be forwarded.

Step 18. In the MLPP alternate party No Answer Ring Duration field, enter the number of seconds that the phone will ring when it receives a precedence call before forwarding to the Forward No Answer destination if unanswered.

Line Settings for all Devices

Step 19. In the Alerting Name field, enter the name that should be displayed on the caller’s phone.

Line Settings for these Devices

Step 20. The set of parameters under Line Settings for this Device define Caller ID, Message Waiting Indicator (MWI) and ring settings. The settings configured here only affect the line on this phone even if it is a shared line appearance. If you want these settings to apply to all phones that have this directory number as a shared line appearance, check the Update Shared Device Settings button next to the setting. The Update Shared Device Settings button is displayed only if the line is shared. The first of these fields, labeled Display (Internal Call ID), is used to configure which caller ID is displayed when calls are placed to other internal callers. Enter up to 30 characters in this field. Both letters and numbers are allowed. If this field is left blank, the lines directory number will be used.

Step 21. The next field, which is labeled Line Text Label, is used to define how the line displays on the phone. If you want the extension number to display next to the line button, leave this field blank. To display a label other than the directory number next to the line button, enter that label in this field.

Step 22. The External Phone Number Mask field may be used to modify the external caller ID for calls placed from this line. An example mask might be 408370XXXX. The extension number is used to fill in the XXXX portion. In this example, if the directory number is 1401, the external phone mask would cause the external caller ID number to be 4083701401. The External Phone Mask on the first line creates the fully qualified directory number that is displayed above the first extension on certain IP Phones. Masks are explored in further detail in Chapter 4: n.

Step 23. The Message Waiting Lamp Policy field which determines if the light on the phone is turned on when a new message is left for this extension. In most cases, this value should be left at Use System Policy. The available choices are as follows:

Light and Prompt— The light turns on and the envelope icon next to the line displays.

Prompt Only— Only the envelope icon next to the line displays.

Light Only— Only the light is turned on.

None— No indication is used.

Use System Policy— Uses the setting selected in CCM services parameters.

Step 24. The next two settings determine if the phone rings when incoming calls are being received on this directory number. In most cases this value should be left at Use System Default. The available choices are as follows:

Disable— Phone does not ring.

Flash Only— The light flashes—No ring.

Ring Once— Rings once and then stops.

Ring— Normal Ringing.

Beep Only— A beep is played (only valid for the Phone Active setting).

Use System Policy— Uses the setting selected under services parameters.

Step 25. Select the desired value for Ring Setting (Phone Idle) and Ring Setting (Phone Active).

Multiple Call / Call Waiting Settings

Step 26. The next field, which is labeled Maximum Number of Calls, determines how many active calls can be on the line. The maximum is 200 active calls per phone. Enter the maximum number of calls in this field. The default of four should be adequate for most phones.

Step 27. The field labeled Busy Trigger determines how many active calls are required before the line is considered busy. The default is two. This means that if the maximum number of calls on the line is four and the busy trigger is two, the third call will receive a busy indication. However, two more calls could be placed from this phone because the maximum number of calls is four.

Forwarded Call Information Display

Step 28. The Forwarded Call Information Display section determines what information is sent when a call is forwarded. Select the information to be sent by checking the box next to each desired field.

Step 29. The last field on this page is the Character Set. Select the character set that is to be used on the display setting for the line from the drop-down list.

Step 30. If this is a new directory number that you have added to the phone, click the Add button at the top of the screen. If you are modifying an existing directory number, click the Update button at the top of the screen.

That’s all there is to it! It really is a pretty simple task once you are familiar with all of the parameters that need to be configured. However, if you are new to CallManager, a number of these parameters may seem confusing. Rest assured that these parameters will be explained in greater detail throughout the remainder of this book.
Using BAT to Add Devices

Imagine having to add more than a hundred phones and you must choose between adding them manually or adding them using auto-registration. At first glance you would most likely choose to add them using auto-registration. Normally, this would be a good choice; however, you discover that each phone needs to have a specific directory number. Then, to make things more complicated, a number of the phones need to use different phone button templates. You could still use auto-registration, but after the phones were added you would have to change the directory numbers and phone button templates of each phone. It would be more efficient to add these phones and not have to go back and touch each one. This is where a utility called Bulk Administration Tool (BAT) comes in. BAT allows you to pre-populate CallManager with all the information for the phones before they are connected to the system. By adding the phones information, a configuration file is created that the phones download from the TFPT server when it boots up.

BAT adds devices to CallManager by importing a Comma Separated Value (CSV) file that contains the required information about the phones. The CSV contains a number of fields that must be populated. The easiest way to create this CSV file is to use an Excel template that is included with BAT. The template offers an easy, well-formatted interface to add all the required and optional information. However, before this template can be used to create a CSV file, BAT must first be installed. BAT is included with CallManager, but is not installed by default. BAT must be installed on the publisher and will cause Internet Information Services (IIS), World Wide Web (WWW) publishing, and File Transfer Protocol (FTP) publishing services to stop during the installation. It is recommended that you install BAT during nonproduction hours, due to the impact that this can have on the server.
Installing BAT

The following steps guide you through the installation process.

Step 1. From within CCMAdmin, select Applications>Install Plugins.

Step 2. From the list of plug-ins on the Install Plug-ins page, click the icon to the left of Cisco Bulk Administration Tool.

Step 3. A window displays asking, “Would you like to open the file or save it to your computer?” Select Open. Depending on the version of Internet Explorer you are using, this message may vary. It may give you the choice, “Run this program from its current location.” If it does, select this option and click OK.

Step 4. The install preparation process begins automatically. During this time you may receive a security warning. If you do, select Yes.

Step 5. A Welcome window displays. Click Next to start the installation.

Step 6. When the installation is complete, a window displays informing you that it is complete. Click the Finish button.

After BAT is installed, the CSV files need to be created. To do this, use the Excel template that was mentioned previously in this section. The number of available fields in the template depends on the model of phone you are importing. For example, the 7960 can contain more than 30 fields. The top row displays what information should be entered into the field and whether it is a required field. The fields include, but are not limited to, information such as the user’s name, MAC address of the phone, directory numbers, and speed dial numbers. When you first open the template, it displays only a few fields. The template then lets you add other fields, as you desire. This feature enables you to avoid dealing with unneeded fields. After the information is added to the template, a CSV file is created and must be placed in the C:\BATFiles\Phones\Insert\directory on the publisher. The necessary steps are presented later in this section.

Although Excel may be used as part of the overall BAT process, it should never be installed on the CallManager. Excel should be loaded on another PC and the file copied to the CallManager after the export to a CSV is completed.

On a large deployment, a Universal Serial Bus (USB) thumb drive may prove useful to transport these files.

It may appear that the hardest part of creating the CSV file is the entry of all the information. However, often what is harder is acquiring the information that must be entered into the template. Make sure you take the proper amount of time beforehand to ensure you have obtained all the required information. The best way to get this information is to perform station reviews. A station review is a formal process that is used to record information about each phone that is to be added to the system. The information acquired during this process includes the number of lines (and directory numbers assigned to each), number of speeds, required features (conference calls, voicemail, and so on) and the type of calls that may be placed by the phone (local, national, international, and so on). This, of course, is only a sample of the information that is gathered during the station review process. It is important that you gather as much information as possible during station reviews. After this information is collected, the creation of the CSV files can begin.

In addition to the CSV file that is created, templates within the BAT application must also be created. These templates define the characteristics of the phones that you are going to add. These templates contain parameters for things such as phone type, phone button template, softkey templates, calling search spaces, number of lines, and partitions just to name a few. After these templates are configured, they are used in concert with the CSV file that has been created to insert phones into the system. Because these templates contain many parameters, it may be necessary to create a number of templates depending upon your environment. For each set of phones that require different settings on any parameter defined within a template, a new template must be created. Let’s look at a simple example so you can see how easily the number of these templates can grow.

Let’s say you have 70 7940s and 50 7960s to deploy. Right off the bat (no pun intended), you know you need at least two templates because there are two types of phones. For this example, let’s assume half the 7940s are going to be two-line phones and the other half are going to be one-line phones. This means that at least two templates are needed for the 7940s alone. As for the 7960s, let’s assume that through the station review process, you determine that you need two-, three-, and four-line phones, which means that at least three templates are needed for the 7960s. If all the phones with the same number of lines do not differ in any other way, five templates in all are needed. However, just to make things interesting, let’s say that half the three-line phones and half of the four-line phones require different calling search spaces. This adds the need for two more templates, bringing the total to seven. You can see how minor changes can cause the number of required templates to grow quite quickly. It is important to understand that because each template is used with a CSV file, you need to create the same number of CSV files as there are templates.

Make sure you take time to determine the number of templates that will be needed. Sometimes after determining how many templates are needed, you may feel it is not worth the effort. I can remember an installation of 3000 phones that required more than 230 templates. Although this was a large number of templates, days if not weeks of time was saved on that deployment by creating the templates and CSV files. Keep in mind that this is an extreme example. Another deployment of 300 phones required only one template.

Before you can determine how many templates you need, you must be familiar with all of the parameters that are set in the BAT templates. Each model of phone has unique parameters, but there are also a number of parameters that are common among all models. The following is a list of parameters that can be defined in the template for most models of phones.

* Device Pool
* Calling Search Space
* AAR Calling Search Space
* User and Network Hold Audio Source
* Location
* User Locale
* Network Locale
* Phone Button Template
* Phone Load Name
* External Data Locations
* Multilevel Precedence and Preemption Information
* Number of Lines

By taking the number of different models of phone that will be deployed and determining how many of the parameters defined in the template will be different among like models, you can determine approximately how many templates and CSV files that must be created.

After you have set up a template, you can often reuse it over and over with only minor modifications.

Now that you have a good overview of what is required for BAT to work, let’s take a look at the steps required to add phones to the system using BAT.

After BAT is installed, you can begin the process of creating the CSV files and the templates. Before creating the templates and CSV files you should have conducted a detailed analysis of the phones that are to be deployed. Based on the disparity of the phones in your system, you should have a good idea of how many templates and CSV files you need. The analysis should also have supplied the information that will be needed to create the templates and files.

You can create either the template or the CSV first. In the steps that follow the creation of the CSV is described first.

It’s a good idea to use same naming convention for both the template and the CSV file that will be used together. This way, when you are selecting the CSV file, it is easy to determine which file goes with which template.
Creating a CSV File for BAT

Step 1. Copy the Excel BAT template to a PC on which Excel is installed. The Excel BAT template can be found in C:\CiscoWebs\BAT\ExcelTemplate on the publisher. The name of this file is bat.xlt.

Step 2. Open the BAT template in Excel. You may get a security warning about macros. Macros must be enabled for this template to function properly, so you may need to change Excel’s settings to allow this.

Step 3. A template like that shown in Figure 3-8 displays. As you can see, by default, only three fields display in the template. Click the Create File Format button to add fields.

03fig08.gif

Figure 3-8 Excel BAT Template
Step 4. A screen similar to that shown in Figure 3-9 should display. Select the device and line fields you want added to the template by highlighting the field in the box on the left side and clicking the >> arrows.

03fig09.jpg

Figure 3-9 Excel BAT Template File Format
Step 5. To remove device and line fields from the template, highlight the field in the box on the right side and click the << arrows.

Step 6. After you move all the desired fields to the box on the right side, click the Create button. When asked if you want to overwrite the existing CSV format, click Yes.

Step 7. Go to the far right of the template and in the appropriately labeled fields, enter the number of lines and speed dials these phones will have. When you enter information in these fields, the template adds the appropriate fields.

Step 8. At this point the template is ready for the data to be entered.

Step 9. After all the data is entered, click the Export to BAT Format button, which is located at the top far-right portion of the template.

Step 10. You are prompted for a place to store this file. BAT creates a default location and file name. An example is c:\XlsDataFiles\Phones#09072004193840. BAT always selects c:\XlsDataFiles as the default file location. The file name is based in part on the tab you have selected to export, such as Phones, PhonesUsers, and UserDeviceProfiles, followed by the number symbol (#). The second part of the filename is the date and time that the file was exported. Note that in the preceding example, 09072004193840 represents September 7th, 2004 at 19:38 and 40 seconds. If you prefer a different file location or name, you can change it before clicking OK.

Step 11. Copy this file to the C:\BATFiles\Phones\Insert\ directory on the publisher.

Adding Phones Using BAT

The process of inserting phones using BAT is actually a four-stage process, and each stage has its own set of steps. The first stage of adding phones in BAT is creating the templates. The following steps walk you through this stage. A 7960 template is created in this example.

Step 1. At the console of the Publisher, navigate to Start>Programs>CallManager 4.x>Bulk Admin Tool>BAT x.x (x.x is the version number).

Step 2. From within the Bulk Administration Tool select Configure>Phones.

Step 3. From the Phone Options screen, make sure the Insert Phones radio button is selected and click the Next button at the bottom of the screen.

Step 4. The Steps to Insert Phones page appears and shows all the steps that must be completed to insert phones. In all, there are four steps, but each step includes additional substeps. Make sure the radio button labeled Step 1: Add, view, or modify phone templates is checked and click the Next button.

Step 5. A screen similar to that shown in Figure 3-10 displays. On this screen you define the parameters for this template. In the Phone Template Name field, enter a name for this template. Remember to try to keep the naming convention the same as the CSV file that will be used with this template.

03fig10.jpg

Figure 3-10 BAT Phone Template Configuration
Step 6. From the drop-down list in the Device Type field, select the type of phone for this template. When a phone is selected, the page refreshes so that all applicable fields are displayed.

Step 7. The remainder of this page contains all the parameters that can be configured for this device. Because all these parameters are covered earlier in this chapter, they are not discussed in this section. If necessary, refer back to steps 6-26 in the Manually Adding Phones section in this chapter for details of each parameter. Enter the appropriate information in each field.

Step 8. After the parameters have been configured, click the Insert button at the top of the screen. An alert window informs you if the operation is successful. Click the OK button in the alert window.

Step 9. After the template is inserted, a Line Details section is added at bottom of the screen. Find this section and click Add Line 1 to define the parameters for this line.

Step 10. A screen similar to that shown in Figure 3-11 displays. This screen is used to define all the parameters for this line. Because all these parameters are covered earlier in this chapter, they are not discussed in this section. If necessary, refer back to steps 6-25 in the Add a line to a Phone section in this chapter for details on each parameter. Enter the appropriate information in each field.

03fig11.jpg

Figure 3-11 BAT Phone Line Configuration
Step 11. Enter the desired information in these fields and click the Insert and Close button. Repeat steps 9-10 for each additional line.

Step 12. When configuring templates for certain models of phones, you may also add speed dial, and services, to the template. To add speed dials to the template, click the Add/Update Speed Dials link in the upper-right corner on the Phone Template Configuration page.

Step 13. A screen displays that allows you to add two types of speed dials. The first type is associated with a button on the phones. The user accesses the second type by pressing the two-digit speed dial number, which has the number they want to reach assigned to it, and by pressing the AbbrDial softkey. Enter the desired speed dials and click Update and Close.

Step 14. To add services to the template, click the Subscribe/Unsubscribe Services link in the upper-right corner on the Phone Template Configuration page. Services are cover in more detail in Chapter 6: Configuring CallManager Features and Services.

Step 15. A window displays that allows you to select the available services. Select the desired service and click Continue. Enter any additional information the service may require.

Step 16. After all services have been added, the template is complete. Click the Back button at the bottom of the page to return to the Steps to Insert Phones page.

The previous stage left you at the page labeled Steps to Insert Phones, on which you should see four options. Step 1: Add, view, or modify phone templates has just been completed, and if you created the CSV data file using the Excel template Step 2: Create the CSV file is also completed. This brings you to Step 3: Validate phone records. This part of the process checks the CSV file that you created to make sure it is formatted properly.

Step 1. On the Step to Insert Phones page, select the radio button that is labeled Step 3: Validate phone records and click the Next button.

Step 2. From the File Name drop-down list, select the CSV file that you created using the Excel template. Only CSV files that you placed in C:\BATFiles\Phones\Insert\ on the publisher display in this list.

Step 3. Select the corresponding template that you created from the Phone Template Name drop-down list.

Step 4. Click the Validate button.

Step 5. After the file has been validated, the status line should state Validate Completed. Click the View Latest Log File link next to the Validate button to view the log.

Step 6. If no errors are reported, close the log window, click the Back button, and move on to the next set of steps.

The last stage is to actually insert the phones into the CallManager database.

Step 1. On the Step to Insert Phones page, select the radio button labeled Step 4: Insert Phones, and click the Next button.

Step 2. From the File Name drop-down list, select the CSV file that you created using the Excel template. Only CSV files that you placed in C:\BATFiles\Phones\Insert\ on the publisher display in this list.

Step 3. Select the corresponding template that you created from the Phone Template Name drop-down list.

Step 4. Click the Insert button.

Step 5. A warning message displays informing you that you are about to insert records. Click OK to continue.

Step 6. While the insertion is being made, a status window displays. After the insertion is completed, the status line should state “Insert Completed.” Click the View Latest Log File link next to the Insert button to view the log.

Step 7. If no errors are reported, close the log window and exit BAT.

Adding Phones Using TAPS

A more advanced method of adding phones called Tool for Auto-registered Phone Support (TAPS) is also available. This tool allows you to pre-populate CallManager’s database with all the information about the phones, except for the MAC address. After the information is entered, the phones are plugged into the system and auto-registered. Because the MAC addresses have not been entered into the system, the TFTP server cannot deliver a con-figuration file to the phone based on the MAC address. This causes a default configuration to be issued and the phone to be registered with basic configuration. After the phone is registered, a predetermined number is dialed that routes the call to an Interactive Voice Response (IVR) server. The IVR server asks what language you would like to use and the extension number that should be assigned to the phone. After this is completed, the MAC address of the phone is associated with the configuration that matches the extension number entered, and a configuration file is created. The phone gets this file from the TFTP server when it reboots.

Integrators often use this method of adding phones during large deployments. Because this method allows the integrator to populate the system without having to add MAC addresses, the deployment of the phones is simpler. Integrators don’t have to make sure a certain phone with a certain MAC address is plugged into a certain jack. The integrator need only be concerned that the proper model is placed in each location. Previous to the introduction of this tool, the MAC addresses had to be entered before the phones were deployed, and this required the integrator to label the box of each phone with the extension number for which it was configured. This also meant that no information could be added to the system until the phones where in hand.

The detailed steps for TAPS are outside of the scope of this book, but further information can be found in Cisco’s Bulk Administration Tool documentation on the Cisco website. You can find the latest copy of this by searching “bulk administration tool guide” at Cisco.com.

More info:Configuring CallManager and Unity: A Step-by-Step Guide (Networking Technology)
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* Read new content on QSIG, SIP trunks, video support, hunt lists and line groups, time-of-day routing, and new features added in CallManager releases 3.2, 3.3, 3.4, 4.0, and 4.1
* Review content from the first edition that has been fully revised and updated to CallManager release 4.1
* Learn how to deploy and manage a CallManager solution
* Understand the components that make up CallManager call routing through the use of basic to advanced examples that solve enterprise call routing problems
* Gain a full understanding of how CallManager manages media resources and processes information for conferencing, transcoding, annunciation, and more
* Learn detailed information about North American and international dial plans, trunk and station devices, media resources, and much more

Cisco CallManager Fundamentals, Second Edition, provides examples and reference information about Cisco® CallManager, the call-processing component of the Cisco IP Communications solution. Cisco CallManager Fundamentals uses examples and architectural descriptions to explain how CallManager processes calls. This book details the inner workings of CallManager so that those responsible for designing and maintaining a Voice over IP (VoIP) solution from Cisco Systems® can understand the role each component plays and how they interrelate.

This book is the perfect resource to supplement your understanding of CallManager. You’ll learn detailed information about dial plan management and call routing, hardware and software components, media processing, call detail records, system management and monitoring, and the history of CallManager. The authors, all members of the original team that developed and implemented the CallManager software and documentation from its early stages, also provide a list of features and Cisco solutions that integrate with CallManager.

This second edition of Cisco CallManager Fundamentals covers software releases through release 4.1. With this book, you will gain a deeper understanding of the system and find answers to questions not available in any other source.

This book is part of the Cisco Press® Fundamentals Series. Books in this series introduce networking professionals to new networking technologies, covering network topologies, example deployment concepts, protocols, and management techniques.

A Cisco IP Communications network is a suite of components that includes Internet Protocol (IP) telephony communications. Cisco CallManager is a core component of a Cisco IP Communications network, the primary function of which is to serve as the call routing and signaling component for IP telephony.

The term IP telephony describes telephone systems that place calls over the same type of data network that makes up the Internet. Although strictly speaking, IP telephony primarily enables users to have voice conversations, CallManager also has the capability to enable users with PCs associated with their phones, users with video-only endpoints, and users with H.323-based video systems to have end-to-end video conversations.

Telephone systems have been around for more than 100 years. Small, medium, and large businesses use them to provide voice communications between employees within the business and to customers outside the business. The public telephone system itself is a very large network of interconnected telephone systems.

What makes IP telephony systems in general, and CallManager in particular, different is that they place calls over a computer network. The phones that CallManager controls plug directly into the same IP network as your PC, rather than into a phone jack connected to a telephones-only network.

Phone calls placed over an IP network differ fundamentally from those placed over a traditional telephone network. To understand how IP calls differ, you must first understand how a traditional telephone network works.

In many ways, traditional telephone networks have advanced enormously since Alexander Graham Bell invented the first telephone in 1876. Fundamentally, the traditional telephone network is about connecting a long, dedicated circuit between two telephones.

Traditional telephone networks fall into the following four categories:

* Key systems
* Private Branch Exchanges (PBX)
* Class 5 switches
* Class 1 to 4 switches

A key system is a small-scale telephone system designed to handle telephone communications for a small office of 1 to 25 users. Key systems can be either analog, which means they use the same 100-year-old technology of your home phone, or digital, which means they use the 30-year-old technology of a standard office phone.

A PBX is a corporate telephone office system. These systems scale from the small office of 20 people to large campuses (and distributed sites) of 30,000 people. However, because of the nature of the typical circuit-based architecture, no PBX vendor manufactures a single system that scales throughout the entire range. Customers must replace major portions of their infrastructure if they grow past their PBX limits.

A Class 5 switch is a national telephone system operated by a local telephone company (called a local exchange carrier [LEC]). These systems scale from about 2000 to 100,000 users and serve the public at large.

Long distance companies and national carriers (called interexchange carriers [IEC or IXC]) use Class 1 to 4 switches. They process truly mammoth levels of calls and connect calls from one Class 5 switch to another.

Despite the large disparity in the number of users supported by these types of traditional networks, the core technology is circuit-based. Consider an old-time telephone operator. He or she sits in front of a large plugboard with hundreds of metal sockets and plugs. (Figure 1-1 shows a picture of an early PBX.) When a subscriber goes off-hook, a light illuminates on the plugboard. The operator plugs in the headset and requests the number of the party from the caller. After getting the number of the called party and finding the called party’s socket, the operator checks to see whether the called party is busy. If the called party is not busy, the operator connects the sockets of the calling and called parties with a call cable, thus completing a circuit between them. The circuit provides a conduit for the conversation between the caller and the called party.
01fig01.jpg

Figure 1-1 An Early PBX

Today’s central switching office—specifically, its call processing software—is simply a computerized replacement for the old-time telephone operator. Obeying a complex script of rules, the call processing software directs the collection of the number of the called party, looks for the circuit dedicated to the called party, checks to see whether the line is busy, and then completes the circuit between the calling and called parties.

In the past, this circuit was an analog circuit from end to end. The voice energy of the speaker was converted into an electrical wave that traveled to the listener, where it was converted back again into a sound wave. Even today, the vast majority of residential telephone users still have an analog circuit that runs from their phone to the phone company’s central switching office, whereas digital circuits run between central switching offices.

This reliance on circuits characterizes traditional telephone systems and gives rise to the term circuit switching. A characteristic of circuit switching is that after the telephone system collects the number of the called party, and establishes the circuit from the calling party to the called party, this circuit is dedicated to the conversation between those calling and called parties. The resources allocated to the conversation cannot be reused for other purposes, even if the calling and called parties are silent on the call. Furthermore, if something happens to disrupt the circuit between the calling and called parties, they can no longer communicate.

Like the central switching office, CallManager is a computerized replacement for a human operator. CallManager, however, relies on packet switching to transmit conversations. Packet switching is the mechanism by which data is transmitted through the Internet, which encapsulates packets according to the Internet Protocol (IP). Web pages, e-mail, and instant messaging are all conveyed through the fabric of the Internet by packet switching. The term voice over IP (VoIP) specifically refers to the use of packet switching using IP to establish voice communications between IP-enabled endpoints on LANs and IP WANs, as well as the Internet (although CallManager is generally not deployed in configurations that route voice traffic over the Internet).

In packet switching, information to be conveyed is digitally encoded and broken down into small units called packets. Each packet consists of a header section and the encoded information. Among the pieces of header information is the network address of the recipient of the information. Packets are then placed on a router-connected network. Each router looks at the address information in each packet and decides where to send the packet. The recipient of the information can then reassemble the packets and convert the encoded data back into the original information.

Packet switching is more resilient to network problems than circuit switching because each packet contains the network address of the recipient. If something happens to the connection between two routers, a router with a redundant connection can forward the information to a secondary router, which in turn looks at the address of the recipient and determines how to reach it. Furthermore, if the sender and recipient are not communicating, the resources of the network are available to other users of the network.

In circuit-switched voice communications, an entire circuit is consumed when a conversation is established between two people. The system encodes the voice in a variety of manners, but the standard for voice encoding in the circuit-switched world is pulse code modulation (PCM). Because PCM is the de facto standard for voice communications in the circuit-switched world, it comes as no surprise that a single voice circuit has been defined as the amount of bandwidth required to carry a single PCM-encoded voice stream.

Video communications require that significantly more information be sent from one end of a connection to another. In circuit-switched video communications, multiple circuits are usually simultaneously reserved for a single call to allow endpoints to exchange high-quality video.

An interesting complication involving voice encoding is introduced by packet-switched communications. Even if circuit-switched systems encode the voice stream according to a more efficient scheme, little incentive exists to do so, because, in most instances, a circuit is fully reserved no matter how little data you place on it. In the packet-switched world, however, a more efficient encoding scheme means that for the same amount of voice traffic, you can place smaller packets on the network, which in turn means that the same network can carry a larger number of conversations. As a result, the packet-switched world has given rise to several different encoding schemes called codecs.

Different types of voice encoding offer different benefits, but generally the more high fidelity the voice quality, the more bandwidth the resulting media stream requires. As the amount of bandwidth that you are willing to permit the voice stream to consume decreases, the more clever and complex the codec must become to maintain voice quality. The codecs that attempt to minimize the bandwidth required for a voice stream require complex mathematical calculations that attempt to predict in advance information about the volume and frequency level of an utterance. Such codecs are highly optimized for the spoken voice. Furthermore, these calculations are often so computationally intensive that software cannot perform them quickly enough; only specialized hardware with digital signal processors (DSP) can handle the computations efficiently. As a result, codec support often differs substantially from device to device in the VoIP network, because devices that do not incorporate DSPs can generally support only easy-to-encode and easy-to-decode codecs such as G.711.

Because not all network devices understand all codecs, an important part of establishing a packet voice call is the negotiation of a voice codec to be used for the conversation. This codec negotiation is a part of a packet-switched call that does not assume nearly the same importance on a circuit-switched call. Chapter 5, “Media Processing,” discusses codecs in more detail.

The information contained in a video call is also encoded using a particular codec; unlike voice codecs, however, of which a handful of variants must be interworked, for interactive videoconferencing, video in the IP world has widely adopted H.263 to encode end-to-end video information (although most products are moving toward H.264).

The rest of this chapter discusses the following topics:

* Circuit-switched systems
* Cisco IP Communications networks
* Enterprise deployment of CallManager clusters

Circuit-Switched Systems

A circuit-switched system is typically a vertically integrated, monolithic computer system. A mainframe cabinet houses a proprietary processor, often along with a redundant processor, which in turn is connected with a bus to cabinets containing switch cards, line cards, and trunk cards.

Line cards control station devices (usually phones), and trunk cards control trunk devices (connections to other telephone systems). A wire runs from a station into a line card and carries both the call signaling and the encoded voice of the station device. Similarly, wires called trunks connect circuit-switched systems together with trunk cards. Line and trunk cards forward received call signaling to the call processing software, while the encoded media is available to the switch cards. Figure 1-2 demonstrates this architecture.
01fig02.gif

Figure 1-2 Traditional Circuit-Switched Architecture
Call Establishment in a Circuit-Switched Telephone System

Call establishment with a circuit-switched system consists of two phases: a session establishment phase and a media exchange phase.

The session establishment phase is the phase in which the telephone system attempts to establish a conversation. During this phase, the telephone system finds out that the caller wants to talk to someone, locates and alerts the called party, and waits for the called party to accept the call. As part of the call establishment, the telephone system also establishes a circuit back from the telephone switch closest to the caller to the caller itself. This circuit permits the caller to hear a ringback tone in the earpiece of the handset and also ensures that, if the called party answers, the end-to-end circuit can be connected as quickly as possible. This optimization eliminates clipping, a condition that occurs when the called party speaks before the circuit is completely formed, causing the caller to miss the initial utterance.

As soon as the telephone system determines that the called party wants to take the call, it completes the end-to-end circuit between the caller and called user, which permits them to begin the media exchange phase. The media exchange phase is the phase in which the endpoints actually converse over the connection that the session establishment phase forges.

Session establishment is the purview of call signaling protocols. Call signaling protocol is just a fancy term for the methods that coordinate the events required for a caller to tell the network to place a call, provide the telephone number of the destination, ring the destination, and connect the circuits when the destination answers. The following represent just a sample of the dozens of call signaling protocols:

* Rudimentary indications that can be provided over analog interfaces
* Proprietary digital methods
* Various versions of ISDN Basic Rate Interface (BRI), which are implementations of ITU-T Q.931
* Various versions of ISDN Primary Rate Interface (PRI), which are also implementations of ITU-T Q.931
* Integrated Services User Part (ISUP), which is part of Signaling System 7 (SS7)

All of these protocols serve the purpose of coordinating the establishment of a communications session between calling and called users.

As part of the session establishment phase, the telephone network reserves and connects circuits from the caller to the called user. Circuit-switched systems establish circuits with commands to their switch cards. Switch cards are responsible for bridging the media from one line or trunk card to another card in response to directives from the call processing software.

After a circuit-switched system forges an end-to-end connection, the end devices (also called endpoints) can begin the media exchange phase. In the media exchange phase, the endpoints encode the spoken word into a data stream. By virtue of the circuit connection, a data stream encoded by one endpoint travels to the other endpoint, which decodes it.

One feature to note is that in a circuit-switched system, the telephone network’s switches are directly involved in both the call signaling and the media exchange. The telephone system must process the events from the caller and called user as part of the session establishment, and then it issues commands to its switch cards to bridge the media. Both the call signaling and the media follow the same path.

Call signaling protocols sometimes embed information about the voice-encoding method to be used to ensure that the endpoints communicate using a common encoding scheme. For voice communications, however, this media negotiation does not assume the importance it does in a packet-based system, in which endpoints generally have more voice-encoding schemes from which to choose.

In summary, a circuit-switched system goes through the following steps (abstracted for clarity) to establish a call:

1. Call signaling—Using events received from the line and trunk cards, the telephone system detects an off-hook event and dialed digits from the caller, uses the dialed digits to locate a destination, establishes a circuit between a ringback tone generator and the caller, offers the call to the called user, and waits for the called user to answer. When the called user answers, the telephone system fully connects a circuit between the caller and called user.
2. Media exchange—By virtue of their connected circuit, the calling and called users can converse. The calling user’s phone encodes the caller’s speech into a data stream. The switch cards in the telephone system forward the data stream along the circuit until the called user’s phone receives and decodes it. Both the call signaling and the media follow a nearly identical path.

Cisco IP Communications Networks

A Cisco IP Communications network is a packet-based system. CallManager is a member of a class of systems called softswitches. In a softswitch-based system, the call signaling components and device controllers are not separated by a hardware bus running a proprietary protocol but instead are separate boxes connected over an IP network and talking through open and standards-based protocols.

CallManager provides the overall framework for communication within the corporate enterprise environment. CallManager handles the signaling for calls within the network and calls that originate or terminate outside the enterprise network. In addition to call signaling, CallManager provides call feature capabilities, the capability for voice mail interaction, and an application programming interface (API) for applications. Among such applications are Cisco Unity, Cisco IP Communicator, Cisco IP Contact Center (IPCC) Enterprise edition, Cisco CallManager Attendant Console, Cisco IP Manager Assistant (IPMA), Cisco Emergency Responder (CER), Cisco Personal Assistant (PA), Cisco MeetingPlace, Cisco IP Queue Manager, and a variety of third-party applications.

A Cisco IP Communications network is by nature more open and distributed than a traditional telephone system. It consists of a set number of servers that maintain static provisioned information, provide initialization, and process calls on behalf of a larger number of client devices. Servers cooperate with each other in a manner termed clustering, which presents administrators with a single point of provisioning, offers users the illusion that their calls are all being served by the same CallManager node, and enables the system to scale and provide reliability.

The remainder of this section discusses the following topics:

* “CallManager History” presents a short history of CallManager.
* “Cisco-Certified Servers for Running Cisco IP Communications” describes the Windows 2000 servers that CallManager runs on.
* “Windows 2000 Services and Tomcat Services on Cisco IP Communications Servers” presents the services that run on the server devices in a Cisco IP Communications network.
* “Client Devices That CallManager Supports” presents the station, trunking, and media devices that CallManager supports.
* “Call Establishment in a Cisco IP Communications Network” describes how a Cisco IP Communications system places telephone calls.
* “Cisco IP Communications Clustering” describes the concept of clustering servers in a Cisco IP Communications system.

CallManager History

There have been several releases of the software that would become CallManager release 4.1. It started in 1994 as a point-to-point video product, but it was recast as an IP-based telephony system in 1997. By 2004, CallManager could support, via multiple clusters, hundreds of thousands of users with a full suite of enterprise-class features.
1994—Multimedia Manager

The application that would become CallManager release 4.1 began in 1994 as Multimedia Manager 1.0. Multimedia Manager was the signaling controller for a point-to-point video product. Multimedia Manager was developed under HP-UX in the language SDL-88.

Specification and Description Language (SDL) is an International Telecommunication Union (ITU)-standard (Z.100) graphical and textual language that many telecommunications specifications use to describe their protocols. An SDL system consists of many independent state machines, which communicate with other state machines solely through message passing and are thus object-oriented. Furthermore, because SDL is specifically designed for the modeling of real-time behavior, it is extremely suitable for call processing software.

Although Multimedia Manager 1.0 was developed in HP-UX, it was produced to run on Microsoft Windows NT 3.51. Each Multimedia Manager server served only as a call signaling source and destination. Multimedia Manager 1.0 managed connections by sending commands to network hubs, which contained the matrix for the video connections. Each hub contained 12 hybrid Ethernet/time-division multiplexing (TDM) ports. Each port could serve either a PC running videoconferencing software or a subhub that managed four PRI interfaces for calls across the public network. In addition, hubs could be chained together using hybrid Ethernet/TDM trunks. At that point in time, the software was somewhat of a hybrid system; Multimedia Manager, running on a Microsoft Windows NT Server 3.51, handled the call signaling and media control over IP like a softswitch, but the media connections were still essentially circuit-based in the network hubs.

Figure 1-3 depicts CallManager as it existed in 1994.
01fig03.gif

Figure 1-3 CallManager in 1994
1997—Selsius-CallManager

Although Multimedia Manager 1.0 worked wonderfully, by 1997 it was clear that Multimedia Manager was not succeeding in the marketplace. Customers were reluctant to replace their Ethernet-only network infrastructure with the hybrid Ethernet/TDM hubs required to switch the bandwidth-hungry video applications. At that point, Multimedia Manager 1.0 changed from a videoconferencing solution to a system designed to route voice calls over an IP network. Unlike the hybrid solution, which required intervening hubs to connect a virtual circuit between endpoints, media signaling traveled over the IP infrastructure directly from station to station. In other words, the system became a packet-switched telephone system.

The change required the development of IP phones and IP gateways. The database, which had been a software application running under Windows NT, became a set of web pages connected to a Microsoft Access database. The new interface permitted administrators to modify the network configuration from any remote machine’s web browser.

The call processing software changed, too. It incorporated new code to control the IP phones and gateways. For this purpose, the Skinny Client Control Protocol (SCCP) and Skinny Gateway Control Protocol (SGCP) were invented. In addition, the software supported Microsoft NetMeeting, an application that uses the H.323 protocol to support PC-to-PC packet voice calls.

At the same time, the call processing software had finally outgrown the SDL development tools. To ensure that the code base could continue to grow, the pure SDL code was converted into an SDL application engine based on C++ that duplicated all of the benefits that the previous pure SDL environment had provided.

Selsius-CallManager 1.0 was born. It permitted SCCP station-to-station and station-to-trunk calls. Each Selsius-CallManager supported 200-feature phones with features such as transfer and call forward.

Figure 1-4 depicts CallManager as it existed in 1997.
01fig04.gif

Figure 1-4 CallManager in 1997
2000—Cisco CallManager Release 3.0

CallManager received a great deal of attention from the marketplace. By 1998, Selsius-CallManager 2.0 had been released, and Cisco Systems, Inc., had become interested in the potential of the product.

After acquiring the CallManager product as a result of its acquisition of Selsius Systems in 1998, Cisco concentrated on enhancing the product. Cisco also simultaneously undertook a huge design and re-engineering effort to provide both scalability and redundancy to the system. Clustering was introduced, and the SDL engine became the Signal Distribution Layer (SDL) engine, which permits the sending of signals directly from one CallManager to another. A redundancy scheme allowed stations to connect to any CallManager in a cluster and operate as if they were connected to their primary CallManager. Support for Media Gateway Control Protocol (MGCP) was added, as was the Cisco IP Phones 7910, 7940, and 7960, which provided a large display, softkeys (virtual buttons on the phone’s display), and access to voice mail, phone settings, network directories, and services.

By mid-2000, Cisco CallManager release 3.0 was complete. It permitted feature-rich calls between H.323 stations and gateways, MGCP gateways, and SCCP stations and gateways. Each cluster supported up to 10,000 endpoints, and multiple cluster configurations permitted the configuration of up to 100,000 endpoints.

Figure 1-5 depicts CallManager release 3.0.
01fig05.gif

Figure 1-5 CallManager in 2000
2001—Cisco CallManager Release 3.1

CallManager release 3.1 built on the foundation of CallManager 3.0. The platform supported more gateway devices and station devices, added enhancements to serviceability, and added more features. Among the specific enhancements were the following:

* Music on hold (MOH)
* Media resource devices available to the cluster, rather than to individual CallManager servers
* Support for digital interfaces on MGCP gateways
* Call preservation between IP phones and MGCP gateways on server failure
* Database support for third-party devices
* Extension mobility
* ISDN overlap sending and T1-CAS support in a variety of VoIP gateways
* Support for Extensible Markup Language (XML) and HTML applications in Cisco IP Phones
* Support for telephony applications through Telephony Application Programming Interface (TAPI) and Java TAPI (JTAPI) and JTAPI/TAPI call processing redundancy support

2001—Cisco CallManager Release 3.2

CallManager 3.2 was a small-scale release that improved the following areas:

* Scalability— Improvements to support up to 20,000 IP phone endpoints per cluster, to improve the number of simultaneous H.323 calls, and to permit CallManager to simultaneously connect to multiple voice messaging systems
* Language— Localization of end-user–visible interfaces, such as phones, end-user applications, gateways, and user-accessible configuration pages to U.K. English and many non-English language and tone sets
* Supported devices— Support for station-oriented analog gateways such as the VG224 and VG248, as well as the Cisco IP Phone 7905
* Features— Auto-answer at destination IP phone for hands-free intercom service, Automated Alternate Routing (AAR) to route calls over the Public Switched Telephone Network (PSTN) when network bandwidth is no longer available, the ability to drop the most recently joined conference participant from an Ad Hoc conference, consultation transfer from applications, and message waiting enhancements

2002—Cisco CallManager Release 3.3

Like CallManager 3.2, CallManager 3.3 was a reasonably small-scale release, but which improved the following areas:

* Scalability— Improvements to support up to 30,000 IP phones per cluster and hundreds of thousands of IP phones using multiple clusters with an H.323 gatekeeper
* H.323 support— Improved ability to support H.323 gatekeeper-controlled connections between CallManager nodes and better scalability and redundancy through support for multiple gatekeepers and alternate H.323 gateways
* Application improvements— Support for Cisco IP Manager/Assistant and Cisco Call Back on Busy applications
* QSIG support— Support for basic call and line identification services using QSIG, a protocol designed to foster feature transparency between different PBXs
* Feature improvements— Distinctive ring per line appearance and configurable call waiting tones for consecutive calls

2004—Cisco CallManager Release 4.0

CallManager 4.0 was a large-scale release that focused quite strongly on features. Chief among the feature changes was a fundamental change in the way that Cisco IP Phones could manage calls. Prior to CallManager 4.0, Cisco IP Phones abided by two main restrictions:

* For any given line appearance, a Cisco IP Phone could have at most two calls, of which one could be actively streaming voice.
* When a Cisco IP Phone was actively streaming voice, other Cisco IP Phones that shared a directory number with the active Cisco IP Phone could not place or receive calls on the shared directory number (although their other directory numbers, if any, could be used to place and receive calls).

In CallManager release 4.0, Cisco IP Phones are no longer restricted to at most two calls per line appearance. Instead, the maximum number of calls per line appearance is configurable, although phones are still restricted to at most one actively streaming call. (An exception is models such as the Cisco IP Phone 7905, 7910, and 7912, which lack a display that would permit a user to efficiently manage more than two calls—these devices are still limited to, at most, two calls.)

Furthermore, in CallManager 4.0, devices that share line appearances are no longer restricted from placing and receiving calls if other devices that share the directory number are actively streaming voice on a call. A phone can continue to place and receive calls until it reaches the maximum threshold (up to 200 calls) configured by the system administrator.

In addition to continuing to support end-user features provided by earlier releases (transfer, Ad Hoc conference, Meet-Me conference, drop the last conference party, call park, call pickup, group call pickup, call back on busy, redial, speed dials, and others), CallManager 4.0 added the following features to Cisco IP Phones:

* Call join— Allows a user to select several calls from the same line on a Cisco IP Phone and conference them all at once.
* Direct transfer— Allows a user to select two calls from the same line on a Cisco IP Phone and transfer (connect) them together.
* Barge and cBarge— Allow a user at one IP phone (the “barger”) to automatically conference himself or herself into a call with two other conversing parties, one of which shares a line with the barger. cBarge relies on an external conference bridge resource; barge mixes the voice on IP phones that contain a built-in bridge, namely the Cisco IP Phones 7940, 7941, 7960, 7961, 7970, and 7971.
* Privacy— Allows a user at one Cisco IP Phone to prevent other users who share a line appearance from viewing the connected name and number identification of parties with which he or she is conversing.
* Abbreviated dialing— Permits a user to quickly dial preconfigured numbers by entering a one-or two-digit index code that represents the speed dial number.
* Conference drop any party— Permits a user who has created a conference to select from a list of currently connected parties and drop one from the conference.
* Immediate diversion— Permits a user who is receiving or already conversing on a call to divert the caller to the diverter’s voice mailbox.
* Malicious call identification— Permits a user to press a button on an active or recently terminated call to notify the system administrator (and service provider) that a harassing or threatening call has been received.
* Multilevel Precedence and Preemption— Permits users to preempt lower priority calls already occurring at the called number with calls designated as higher priority. This feature is used primarily by the military.
* Hunt groups— Native hunt group capability in CallManager. Hunt groups support broadcast (ring all members), top down, circular, and longest idle hunting. A ring no answer timer can be applied to determine the time to wait before proceeding to the next point.

You can learn more about these features in Chapter 3, “Station Devices.” Learn more about hunt groups in Chapter 2, “Call Routing.”

In addition to focusing on features, CallManager 4.0 improved the following areas:

* QSIG support— Addition of QSIG supplementary services for call diversion and call transfer to permit display updates when calls across multiple PBXs are transferred or forwarded and to support delivery of message waiting indications between PBXs
* Video support— Addition of media control capabilities to support the establishment of video calls from either video-enabled Cisco IP Phones, third-party SCCP video endpoints, H.323-based video endpoints, and audio-only Cisco IP Phones that have an associated PC for video display
* Security— Support for signaling authentication of Cisco IP Phones to prevent rogue phones from registering with CallManager or impersonating other devices, support for Cisco IP Phone signaling, support for media encryption between phones, and integrated support for multiple levels of administrator access
* SIP support— Addition of the Session Initiation Protocol (SIP) call signaling protocol specifically for connections to phone systems outside of a CallManager cluster

2004—Cisco CallManager Release 4.1

CallManager 4.1 continues to focus on support for new features. The following list summarizes the new additions:

* QSIG enhancements— CallManager continued implementing the QSIG protocol for feature transparency with other PBXs.
o — Path replacement optimizes the path between two parties to remove circuit hairpins that form when one party transfers a call to another party.
o — Call forward by rerouting prevents circuit hairpins from forming when a phone on one system forwards a call to a phone on another system.
o — Call back on no reply and call back on busy allow a caller to set a monitor on a called number that has not accepted a call and receive a prompt to redial the called number when it becomes available.
o — Called name allows a calling user to see the name of the party he or she is calling, even when the called party is served by a different call agent.
* Dialed number analyzer is a service that allows you to enter dial strings on behalf of calling devices and analyze how the call may route.
* Time of day routing provides a flexible mechanism by which you can activate and deactivate route partitions according to a schedule. Chapter 2, describes this feature in more detail.
* Client matter codes allow you to define post-dial strings that are associated with specific clients that users can dial to attribute the cost of the call to the client. The codes show up in call detail records to achieve billing traceability.
* Forced authorization codes allow you to define post-dial strings that users must dial to reach their destinations.

Figure 1-6 depicts CallManager release 4.1.
01fig06.gif

Figure 1-6 CallManager in 2005
Cisco-Certified Servers for Running Cisco IP Communications

CallManager and its associated services run on a Windows 2000 server. Because voice applications are so critical to an enterprise’s function, however, Cisco Systems requires that CallManager be installed only on certified server platforms.

Cisco Systems provides a suite of certified servers called Media Convergence Servers (MCS). In addition to these servers, Cisco allows users to install Cisco IP Communications software on servers offered by HP and IBM. Customer-provided servers must match exact server configurations provided by Cisco, because any deviations from the specifications might result in an incomplete install and an unsupported system.

Note

At times, this book uses the term server and, at other times, it uses the term node, particularly when describing CallManager clustering.

A CallManager cluster consists of networked servers running a variety of services that together provide an enterprise VoIP system. Some of these servers in the cluster are generally dedicated to the CallManager database or TFTP service. Others run CallManager, the call processing component of a Cisco IP Communications system.

This book uses the term node to refer specifically to the servers in a CallManager cluster that are running the CallManager service. It’s not uncommon to read a sentence referencing both nodes and servers. For example, it’s consistent to state both that a CallManager cluster can consist of a maximum of 20 servers and that it can consist of a maximum of 8 nodes, because the 12 non-call processing servers handle services such as the Publisher database, TFTP, Cisco IP Voice Media Streaming App, and applications.

The current list, as of the release of 4.1, of MCSs that Cisco ships are as follows:

* Cisco MCS 7815I-2000
* Cisco MCS 7825H-3000
* Cisco MCS 7835H-3000
* Cisco MCS 7835I-3000
* Cisco MCS 7845H-3000
* Cisco MCS 7845I-3000

In addition to MCS, users can build Cisco IP Communications systems based off of the following HP and IBM platforms. (You can find the latest system information and specific parts lists on Cisco.com at http://www.cisco.com/go/swonly.)

* Compaq DL320-G2 Pentium 4 3060 MHz
* Compaq DL320 Pentium III 800 MHz
* Compaq DL320 Pentium III 1133 MHz
* Compaq DL320-G2 Pentium 4 2.26 GHz
* Compaq DL380 Pentium III 1000 MHz
* Compaq DL380 G2 Pentium III 1266 MHz
* HP DL380-G3 Xeon 3060 MHz
* HP DL380-G3 Dual Xeon 3.06 GHz
* HP DL380-G2 Pentium III 1400 MHz
* HP DL380-G2 Pentium III 1400 MHz
* HP DL380-G3 Xeon 2400 MHz
* HP DL380-G2 Dual Pentium III 1400 MHz
* HP DL380-G3 Dual Xeon 2.4 GHz
* IBM xSeries 306 Single-Processor 3.06 GHz
* IBM xSeries 346 Single-Processor 3.4 GHz
* IBM xSeries 346 Dual-Processor 3.4 GHz

Cisco MCS ships with an installation disk that contains all of the Windows 2000 services that are required to create a working IP telephony network. The HP and IBM servers are hardware-only; you must order a software-only version of CallManager (and the Windows 2000 installation) from Cisco to install on these servers.

Cisco IP Communications consists of a suite of applications that you can provision in numerous ways for flexibility. For example, although a server contains applications for managing the database, device initialization, device control, software conferencing, and voice mail, you might decide to reserve an entire server for just one of these functions in a large, differentiated Cisco IP Communications deployment. Servers that perform a sole function are called dedicated servers. For an overview of the services CallManager supports, see the section “Windows 2000 and Tomcat Services on Cisco IP Communications Servers.”

The following list describes Cisco MCS 7800 series servers (two other servers, the MCS-7855-1500 and MCS-7865I-1500, are Cisco Unity-specific).

* MCS-7815I-2000 server— The only tower system that Cisco ships. It is suitable for smaller installations and can be configured to run CallManager, Unity, or Unity Bridge. This server can support up to 300 Cisco IP Phones. MCS 7815 server can only be deployed in the minimal cluster configuration, with one MCS running a Publisher database, Cisco TFTP, and backup CallManager and with the other MCS handling active call processing services.
* MCS-7825H-3000 server— A rack-mountable system, requiring a single rack space. This system can be configured to run CallManager, Cisco Conference Connection (CCC), Cisco Emergency Responder (CER), Cisco IPCC Express (Integrated Contact Distribution [ICD]), Cisco IP Interactive Voice Response (IP IVR), Cisco Personal Assistant, Cisco Queue Manager, and Cisco Unity Unified Messaging. This server can support up to 1000 IP phones or, via clustering, up to 4000 IP phones.
* MCS-7835H-3000 and MCS-7835I-3000 servers— Rack-mountable systems that require two rack spaces and have a single 3.06-GHz processor. These servers can be configured to run CallManager, Cisco Conference Connection (CCC), Cisco Emergency Responder (CER), Cisco IPCC Express (ICD), Cisco IP Interactive Voice Response (Cisco IP IVR), Cisco Personal Assistant, Cisco Queue Manager, and Cisco Unity Unified Messaging. These servers can support up to 2500 IP phones or, via clustering, up to 10,000 IP phones.
* MCS-7845H-3000 and MCS-7845I-3000 servers— Rack-mountable systems that require two rack spaces and have dual 3.06-GHz processors. These servers can be configured to run CallManager, Cisco Conference Connection (CCC), Cisco Emergency Responder (CER), Cisco Internet Service Node (ISN), Cisco IPCC Express (ICD), Cisco IP Interactive Voice Response (Cisco IP IVR), Cisco Personal Assistant, Cisco Queue Manager, and Cisco Unity Unified Messaging. These servers can support up to 7500 IP phones or, via clustering, up to 30,000 IP phones.

Windows 2000 and Tomcat Services on Cisco IP Communications Servers

Cisco IP Communications relies on several Windows 2000 services, of which Cisco CallManager is only one. Cisco IP Communications uses the Windows 2000 services described in Table 1-1.
Table 1-1. Windows 2000 Services That Run on a Cisco IP Communications Server

Service

Description

Cisco CallManager

Provides call signaling and media control signaling for up to 7500 devices. You can have up to eight instances of the CallManager service per cluster.

Cisco Certificate Authority Proxy Function

Manages security certificates for Cisco IP Phones such as the Cisco 7940 and 7960 that do not directly support installed certificates.

Cisco CTIManager

Provides support for the TAPI and JTAPI application interfaces.

Cisco IP Voice Media Streaming App

Provides media termination, RFC 2833 tone interworking, inband tone services for SIP, MOH, and G.711 media mixing capabilities.

Cisco Messaging Interface

Permits Simple Message Desk Interface (SMDI) communications to voice messaging systems over an RS-232 connection.

Cisco MOH Audio Translator

Converts any audio file format compatible with DirectShow and converts it to G.711, G.729a, and wideband codec for MOH to IP telephony endpoints.

Cisco RIS Data Collector

Collects serviceability information from all cluster members for improved administration.

Cisco Telephony Call Dispatcher

Allows users such as receptionists and attendants to receive and quickly transfer calls to other users in the organization; provides automated routing capabilities.

Cisco TFTP

Provides preregistration information to devices, including a list of CallManager nodes with which the devices are permitted to register, firmware loads, and device configuration files.

Cisco Database Layer Monitor (provides database notification)

A change notification server and watchdog process that ensures that all Cisco IP Communications applications on a server are working properly.

Publisher database

Serves as the primary read-write data repository for all Cisco IP Communications applications in the cluster. The Publisher database replicates database updates to all Subscriber databases in the cluster.

Subscriber database

Serves as a backup read-only database for Cisco IP Communications applications running on the server, should the applications lose connectivity to the Publisher database.

Cisco CDR Insert

Periodically scans local call detail record (CDR) files logged by CallManager nodes and inserts them into the CDR database.

Cisco CTL Provider

Accepts connections from the CTL Client utility, which allows you to change the cluster security mode and update the cluster’s Certificate Trust List (CTL).

Cisco Extended Functions

Provides the Quality Reporting Tool service, which allows users to report problems with their phone via the QRT softkey.

Cisco Serviceability Reporter

Generates a daily serviceability summary report for the cluster, including server performance, alerts generated by system, call activities, and other information.

While Table 1-1 indicates native Windows 2000 services that provide call-related services, Cisco IP Communications also supports applications that run as Java servlets hosted by the Apache plug-in Tomcat. Table 1-2 lists the Tomcat applications that Cisco IP Communications supports in the 4.1 release.
Table 1-2. Tomcat Applications That Run on a Cisco IP Communications Server

Name

Description

Cisco Web Dialer

Allows corporate directories to support click-to-dial functionality in which a user viewing a directory page can click a link to have his or her IP phone automatically call the selected person.

Cisco IP Manager Assistant

Provides an enhanced suite of services especially suited for managing the relationship between managers and assistants. This suite includes call filtering, immediate diversion, and send all calls functions.

Cisco Extension Mobility

Allows a user at a Cisco IP Phone to provide a user ID and password to log in to the phone and retrieve his or her extension and customized line settings.
Client Devices That CallManager Supports

In a Cisco IP Communications network, CallManager is the telephone operator, and it places calls on behalf of many different endpoint devices. These devices can be classified into the following categories:

* Station devices— Station devices are generally, but not always, telephone sets. CallManager offers a variety of sets, which it controls with SCCP.

Cisco IP Phone 7902 is a cost-effective, single-line, entry-level station with no display.

Cisco IP Phones 7905G and 7912G are single-line, entry-level phones (with a format different from the Cisco IP Phone 7910G) with a graphical display.

Cisco IP Phone 7920 is a mobile 802.11b phone that enables voice communications over wireless LANs.

Cisco IP Phone 7935 and 7936 are console speakerphones with softkey displays designed for use in conference rooms. They do not support inline power and do not have a switch for supporting an associated PC.

Cisco IP Phone 7940G supports two line/feature buttons and offers a nine-line display with softkeys and status lines.

Cisco IP Phone 7941G supports two line/feature buttons with lighted keys and offers a high-resolution display with softkeys and status lines.

Cisco IP Phone 7960G supports six line/feature buttons and has the same display as the Cisco IP Phone 7940G.

Cisco IP Phone 7961G supports up to six line/feature buttons with lighted keys and has the same display as the Cisco IP Phone 7941G.

Cisco IP Phone 7970G offers eight line/feature buttons and an 11-line backlit, high-resolution color display with touch screen and additional softkeys.

Cisco IP Phone 7971G-GE provides unconstrained bandwidth to desktop applica-tions via Gigabit Ethernet (GE) and features eight line/feature buttons and an 11-line backlit, high-resolution color display with touch screen and additional softkeys.

Cisco IP Phone 7914 expansion modules can be added to Cisco IP Phone 7960G. Each expansion module adds 14 buttons and up to 2 modules can be added to a Cisco IP Phone.

Station devices need not be physical handsets. CallManager also supports H.323 user clients, such as the following:
o — NetMeeting, which runs as a software application on a user’s PC
o — Cisco IP Communicator, a software-based phone that connects to CallManager using SCCP
o — Cisco IP SoftPhone, which connects to CallManager using the TAPI application interface

Chapter 3 goes into more detail about station devices.
* Gateway devices— Gateways provide a bridge between two end users whose endpoints utilize different protocols. Gateways allow IP phones to interact with the billions of already deployed phones in the world.

Gateway devices generally provide one of two types of interconnections. One type of interconnection is from one telephone system to another. This access can be from one network of CallManager nodes to another, from a CallManager network to a PBX or from a CallManager network to a public network such as a Class 4 or Class 5 switch. (But note that intercluster H.323 trunks provide an alternative for connecting CallManager networks together without requiring a gateway device.)

Gateways do not necessarily need to provide access to other networks, however. Gateways can also be used to interwork VoIP directly with traditional telephones (POTS phones).

CallManager controls gateways via three protocols: H.323, MGCP, and the legacy Skinny Gateway Control Protocol (SGCP). On their circuit interfaces, gateways provide both digital—for example, BRI, T1/E1 Channel Associated Signaling (CAS) and T1/E1 Primary Rate Interface (PRI)—and analog (the same type of telephone interface that probably runs into your home) interfaces.

Cisco gateways fall into three general categories:
o — Cisco IOS integrated routers are gateways that provide IP routing in addition to their gateway services. These can be viewed as IP routers that just happen to provide support for analog phones or for analog or digital trunk interfaces. Cisco IOS routers accept voice interface cards (VIC) and voice/WAN interface cards (VWIC) that can provide connectivity to the PSTN using many telephony protocols (as well as media services such as transcoding, media termination, and conference mixing).
o — Cisco standalone voice gateways operate solely as end devices; they do not route IP traffic from network to network. The Cisco ATA 186, ATA 188, VG224, and VG248 provide CallManager with gateway services from its IP phones to analog phones or trunks.
o — Cisco Catalyst voice gateway modules also operate solely as end devices. These modules are inserted into the Cisco Catalyst 6xxx chassis. Cisco Catalyst 6xxx can accept the Communication Media Module (CMM), the 6608 module, and the 6624 module. The CMM, in turn, can take port adapters that support the T1, E1, or FXS telephony interfaces.

Chapter 4, “Trunk Devices,” goes into more detail about trunk devices.
* Media processing devices —Media processing devices perform codec conversion, media mixing, and media termination functions. CallManager controls media processing devices using SCCP. Five types of media processing devices exist.
o — Transcoding resources—These exist to perform codec conversions between devices that otherwise could not communicate because they do not encode voice conversations using a common encoding scheme. If CallManager detects that two endpoints cannot interpret each other’s voice-encoding schemes, it inserts a transcoder into the conversation. Transcoders serve as interpreters. When CallManager introduces a transcoder into a conversation, it tells the endpoints in the conversation to send their voice streams to the transcoder instead of to each other. The transcoder translates an incoming voice stream from the codec that the sender uses into the codec that the recipient uses, and then forwards the voice stream to the recipient. The Catalyst 6xxx platform offers a blade that performs transcoding functions and the NM-HDV, NM-HDV2, and NM-HD-2VE modules support transcoding functions for IOS gateways.
o

— Unicast conferencing devices—These exist to permit Ad Hoc and Meet-Me conferencing. When an endpoint wants to start a multiple-party conversation, all the other parties in the conversation need to receive a copy of its voice stream. If several parties are speaking at once in a conversation, some component in the conversation needs to combine the independent voice streams present at a particular instant into a single burst of sound to be played through the telephone handset.

Unicast conferencing devices perform the functions of both copying a conference participant’s voice stream to other participants in the conference and mixing the voice streams into a single stream. When you initiate a conference, CallManager looks for an available Unicast conferencing device and dynamically redirects all participants’ voice streams through the device. The Catalyst 6xxx platform offers a blade that performs mixing functions, and NM-HDV, NM-HDV2, and NM-HD-2VE modules support mixing functions for Cisco IOS gateways. In addition, the Cisco IP Voice Media Streaming App is a software application that can mix media streams encoded according to the G.711 codec.
o

— Media termination point (MTP) resources—These devices exist to allow users to invoke features such as hold and transfer, even when the person they are conversing with is using an H.323 endpoint such as NetMeeting. Devices that are only H.323v1-compatible do not tolerate interruptions in their media sessions very well. Attempts to place these devices on hold will cause them to terminate their active call. A media termination device serves as a proxy for these old H.323 devices and allows them to be placed on hold as part of feature operation.

CallManager also uses MTPs to interwork with SIP networks. SIP networks generally encode DTMF tones directly in the RTP stream using RFC 2833, while CallManager typically encodes tones directly in the signaling stream. An MTP can provide the interworking between these different types of tones as well as provide inband ringback when a Cisco IP Phone transfers a SIP caller.

The Catalyst 6xxx platform offers modules that perform media termination functions, and modules that provide media termination functions also exist for Cisco IOS routers. Furthermore, the Cisco IP Voice Media Streaming App is a software application that can perform media termination functions for calls that use the G.711 codec.
o — Music on Hold (MOH) resources—These exist to provide users a music source when you place them on hold. When you place a user on hold, CallManager renegotiates the media session between the party you place on hold and the MOH device. For as long as you keep the user on hold, the MOH device transmits its audio stream to the held party. When you remove the user from hold, CallManager renegotiates the media stream between your device and the user.
o — Annunciator resources—These exist to provide users audio announcements when error conditions occur such as preemption due to higher-priority calls, invalid dialed digit strings, or other problems CallManager encounters when placing calls.

Table 1-3 provides a comprehensive list of the Cisco IP Phones that CallManager supports.
Table 1-3. Cisco IP Phones That CallManager Supports

Name

Description

Cisco IP Phone 12SP+

Legacy phone with 12 feature buttons and 2-line text display

Cisco IP Phone 30VIP

Legacy phone with 30 feature buttons and 2-line text display

Cisco IP Phone 7902

Single-line appearance phone with no display

Cisco IP Phone 7905G

Single-line appearance phone with 2-line graphical display

Cisco IP Phone 7910G

Legacy single-line appearance phone with 2-line black-and-white alphanumeric display

Cisco IP Phone 7912G

Single-line appearance phone with 2-line graphical display

Cisco IP Phone 7920

6-line appearance wireless LAN phone (802.11b) with 9-line grayscale graphical display

Cisco IP Phone 7935

Speakerphone console with alphanumeric display designed for use in conference rooms

Cisco IP Phone 7940G

Dual-line appearance phone with 9-line grayscale graphical display

Cisco IP Phone 7941G

Lighted button, dual-line appearance phone with high resolution graphical display

Cisco IP Phone 7960G

6-line appearance phone with 9-line grayscale graphical display

Cisco IP Phone 7961G

Lighted button, 6-line appearance phone with high resolution grayscale graphical display

Cisco IP Phone 7970G

Lighted button, 8-line appearance phone with 9-line color graphical touch screen display

Cisco IP Phone 7971G-GE

Gigabit Ethernet lighted button 8-line appearance phone and 9-line color graphical touch screen display

Microsoft NetMeeting

Windows-based H.323 software client application

Cisco IP SoftPhone

Windows-based JTAPI software client application

Cisco IP Communicator

Windows-based SCCP software client application

Table 1-4 provides a list of the gateway devices that CallManager supports.
Table 1-4. Cisco Gateways That CallManager Supports

Gateway Model

Gateway Control Protocol

Trunk Interface

Port Types

Cisco IOS Integrated Routers

Cisco 1750

H.323

FXS

Loop start or ground start

FXO

Cisco 1751

MGCP

FXS

Loop start or ground start

Cisco 1760

H.323

FXO

E&M

SIP

T1/E1 PRI

T1 PRI

T1 CAS

E1 PRI

E1 CAS R2

Cisco 2600 series

MGCP

FXS

Loop start or ground start

Cisco 2800 series

H.323

FXO

T1/E1 PRI

SIP

BRI

E&M

(Only MGCP supports QSIG.)

T1/E1 PRI

(Only H.323 supports E1 CAS R2.)

T1 CAS

E1 CAS R2

QSIG (Not all Cisco 2600 series gateways support QSIG. Refer to your gateway documentation.)

Cisco 3600 series

MGCP

FXS

Loop start or ground start

Cisco 3700 series

H.323

FXO

T1/E1 PRI

Cisco 3800 series

SIP

BRI

E&M

(Only MGCP supports QSIG.)

T1/E1 PRI

T1/E1

(Only H.323 supports E1 CAS R2.)

T1 CAS

PRI

E1 CAS R2

QSIG (Not all Cisco 3600 series gateways support QSIG. Refer to your gateway documentation.)

Cisco 7200 series

MGCP

T1/E1 CAS

T1/E1 CAS

Cisco 7500 series

H.323

T1/E1 PRI

T1/E1 PRI

SIP

QSIG

Cisco AS5300

H.323

T1/E1 CAS

T1/E1 CAS

Cisco AS5350

T1/E1 PRI

T1/E1 PRI

Cisco AS5400

Cisco Standalone Voice Gateways

Cisco Voice Gateway 200 (VG200)

MGCP or H.323

FXO

Loop start or ground start

(Only MGCP supports QSIG.)

FXS

T1/E1 PRI

T1/E1 PRI

E&M

T1 CAS

T1/E1 PRI

QSIG

Cisco Voice Gateway 224 (VG224)

MGCP or SCCP

FXS

FXS

Cisco Access Digital Trunk Gateway DE-30+

MGCP

E1 PRI

E1 PRI

QSIG

E1 PRI

Cisco Access Digital Trunk Gateway DT-24+

MGCP

T1 PRI

T1 PRI

T1 CAS

E&M

FXO

Loop start or ground start

QSIG

T1 PRI

Cisco Access Analog Trunk Gateway (AT-2, AT-4, AT-8)

Skinny Gateway Control Protocol

FXO

Loop start

Cisco Access Analog Station Gateway (AS-2, AS-4, AS-8)

Skinny Gateway Control Protocol

FXS

Loop start

Cisco VG248 Analog Phone Gateway

SCCP

FXS

Loop start

Cisco IAD2420

MGCP

FXS

Loop start or ground start

FXO

T1 PRI

T1 PRI

E&M

T1 CAS

T1 PRI

QSIG

Cisco Catalyst Voice Gateway Modules

Cisco Catalyst 4000 Access Gateway Module (WS-X4604-GWY)

MGCP or H.323

FXS

POTS

(Only MGCP supports QSIG.)

FXO

Loop start or ground start

T1 CAS

E&M

T1/E1 PRI

T1/E1 PRI

QSIG

T1/E1 PRI

Cisco Catalyst 4224 Voice Gateway Switch

MGCP or H.323

FXS

POTS

(Only MGCP supports QSIG.)

FXO

Loop start or ground start

T1/E1 PRI

T1/E1

T1 CAS

PRI

QSIG

E&M

T1/E1 PRI

Cisco Catalyst 6000 8-Port Voice T1/E1 and Services Module

MGCP

T1/E1 PRI

T1/E1 PRI

T1 CAS

E&M, loop start, ground start

QSIG

(WS-X6608-T1)

T1/E1 PRI

(WS-X6608-E1)

Cisco Catalyst 6000 24-Port FXS Analog Interface Module

MGCP

FXS

POTS

(WS-X6624-FXS)

Cisco Communication Media Module

MGCP

FXS

POTS

(WS-X6600-24FXS)

Cisco Communication Media Module

MGCP

T1 PRI

T1 PRI

T1 CAS

E&M

(WS-X6600-24FXS)

E1 PRI

E1 PRI
Call Establishment in a Cisco IP Communications Network

Call establishment between circuit-switched and VoIP systems is more similar than different. While a circuit-switched system relies on a two-phase process that consists of a call signaling phase (into which commands to connect circuits are included) and a media exchange phase, a VoIP system usually deconstructs call establishment into the following three phases:

1. Call signaling—Like a circuit-switched call, VoIP systems need to coordinate the placing, offering, and answering of a call; that is, given a person named Alice who wants to call another person named Bob, the call signaling step answers the question, “Do Alice and Bob want to talk?”
2. Media control—Unlike traditional circuit-switched systems, however, VoIP systems enable the endpoints to talk directly to each other over the IP infrastructure. While a circuit-switched system has a sort of tacit media control phase in which it asks the switching fabric to join two circuits, a VoIP system uses a more robust phase to enable the endpoints in the call to exchange IP and port information so that the endpoints can connect themselves. In this respect, the media control step answers the question, “How should Alice and Bob talk?”

Note

SIP and MGCP combine the exchange of media control information with the call signaling phase, although this behavior doesn’t change the underlying fact that the endpoints are ultimately connecting themselves. H.323 also supports an integrated signaling and media control phase via its optional fast start procedure.
3. Media exchange—After IP and port information has been exchanged, the endpoints encode information into Real-Time Transport Protocol (RTP) or Secure Real-Time Transport Protocol (SRTP) packets, which they stream directly to each other over the IP infrastructure. In the case of CallManager, this means that, although CallManager is handling the call signaling and media control phases, CallManager has nothing to do with the actual exchange of the conversation, which is a function of the phones and the IP routers that connect them. The media exchange phase answers the real important questions—questions such as “How about we go out for pizza Friday?”

Figure 1-7 shows a comparison between the circuit-switched and packet-switched call models.
01fig07.gif

Figure 1-7 Circuit-Switched Call Versus Packet-Switched Call

Phones are connected directly into the circuit-switched system.

Phones connect to CallManager through a network of routers.

1 Call signaling: The system detects a call request and extends the call to the destination. Negotiation of the type of connection usually occurs as part of the call signaling itself.

1 Call signaling: CallManager detects a call request and extends the call to the destination.

2 Media exchange: When the call is answered, the circuit-switched system must bridge the voice stream. Both call signaling and media exchange are centralized.

2 Media control (sometimes, but not always, part of call signaling): When the destination answers, the endpoints must negotiate a codec and exchange addresses for purposes of exchanging media.

3 Media exchange: The phones exchange media directly with each other. The media often follows a completely different set of routers than the call signaling. Call signaling and media control are centrally managed, but the high-bandwidth media is distributed.

Using the IP network as a virtual matrix offers some remarkable benefits. The Internet is an IP network that spans the globe. A computer on the Internet can talk to its neighbor as easily as it talks to a computer located 1000 miles away. Similarly, without the need to connect circuits one leg at a time across long distances, one CallManager can connect calls between IP phones separated by area codes or even country codes as easily as it can connect two IP phones in the same building.

Furthermore, IP networks are distributed by their nature. A traditional circuit-based solution requires that all the wires for your voice network run into the same wiring closet. This means that the telephone system can intercept events from the line and trunk cards and gain access to the media information that the devices send to connect them in the matrix. CallManager can communicate with devices by establishing virtual wires through the fabric of the IP network, and the devices themselves establish virtual wires with each other when they start exchanging media. This feature makes CallManager more scalable than traditional circuit-switched systems. Figure 1-8 offers a comparison.
01fig08.gif

Figure 1-8 Cisco IP Communications (IPC) Scalability

Another major benefit of CallManager is that it resides on the same network as your data applications. The Cisco IP Communications model is a traditional Internet client/server model. CallManager is simply a software application running on your data network with which clients (telephones and gateways) request services using IP interfaces. This co-residency between your voice and data applications allows you to integrate traditional data applications (such as web servers and directories) into the interface of your voice devices. The use of standard Internet protocols for such applications (HTML and XML) means that the skills for developing such applications are readily available, if you want to customize the services available to your voice devices.

Finally, CallManager interacts with IP devices on the network using call signaling protocols, which allows you to mix and match equipment from other vendors when building your voice network. For devices, CallManager supports SCCP to phones, gateways, and transcoding devices; MGCP to gateway devices; H.323 to user and gateway devices; and SIP to other SIP networks. For CTI applications, CallManager supports TAPI and JTAPI.
Cisco IP Communications Clustering

A traditional telephone system tends to come packaged in a large cabinet with racks of outlying cabinets to house the switch cards, line cards, and trunk cards. A Cisco IP Communications network, however, is composed of a larger number of smaller, more specialized components. This allows you to more closely tailor your telephone network to your organization’s needs.

This focus on the combined power of small components extends to CallManager, the call processing component of a Cisco IP Communications network. Within a cluster, up to eight servers can be dedicated to running the CallManager service to handle the call routing, signaling, and media control for the enterprise, with other servers dedicated to providing database services, TFTP, applications, and media services such as conferencing, media termination, music on hold, or annunciation. Such a set of networked servers is called a CallManager cluster. Clustering helps provide the wide scalability of a Cisco IP Communications network, redundancy in the case of network problems, ease of use for administrators, and feature transparency between users.

Clustering allows for flexibility and growth of the network. In release 4.1, clusters can contain up to eight call processing nodes, which together can support 30,000 endpoints. If your network serves a smaller number of users, you can buy fewer servers. (Using multiple clusters served by an H.323 gatekeeper, CallManager can support larger networks—up to hundreds of thousands of phones.) As your network grows, you can simply add more servers. Clustering allows you to expand your network seamlessly.

The idea behind a cluster is that of a virtual telephone system. A cluster allows administrators to provision much of their network from a central point. Cluster cooperation works so effectively that users might not realize that more than one CallManager node handles their calls. A guiding philosophy of clustered operation is that if a user’s primary CallManager node experiences an outage, the user cannot distinguish any change in phone operation when it registers with a secondary or tertiary CallManager. Thus, to the users and the administrators, the individual nodes in the cluster appear as one large telephone system, even if your users reside in completely different geographical regions.

CallManager cluster members do not need to be co-resident. In fact, geographically separating the cluster members can provide even greater device survivability. If a disaster occurs in one geographic site (if, for instance, the CallManager system administrator receives one too many special executive requests and takes a fire ax to the Media Convergence Servers), nodes in other geographic sites can take over the phones. Separating CallManager cluster members in this fashion is called clustering over the WAN.

Clustering over the WAN currently requires a high-performance network between the cluster members. The following list summarizes the guidelines:

* At most a 40-ms round-trip packet delay between any two CallManager nodes
* At most four active CallManager nodes (with four standby CallManager nodes for failover)
* 900 kbps per each 10,000 Busy Hour Call Attempts (BHCA) in the cluster (with more bandwidth required if you want to support device failover across the WAN)

If your network doesn’t meet the guidelines for clustering over the WAN, deployment options are still available to you:

* Remote sites should run independent clusters—a model called distributed call processing.
* Devices in remote sites should be managed by a cluster of servers that reside in a central site, a model called centralized call processing.
* Both the centralized and distributed models should be used in a combined model.

Large networks tend to deploy a combination of distributed and centralized call processing systems.

Because performance characteristics and supported deployment models change from release to release, be sure to check http://www.cisco.com/go/srnd for current models and additional information.
Clustering and Reliability

Clustering provides for high reliability of a Cisco IP Communications network. In a traditional telephone network, a fixed association exists between a telephone and the call processing software that serves it. Traditional telephone vendors provide reliability through the use of redundant components installed in the same chassis. Table 1-5 draws a comparison between a traditional telephone system’s redundant components and Cisco IP Communications redundancy.
Table 1-5. Comparison Between Traditional Telephone System Redundancy and Cisco IP Communications Redundancy

Function

PBX

Cisco IP Communications

Processor unit

Redundant

Up to eight call processing nodes (running CallManager) with one Publisher database, up to two TFTP servers, and other application and media servers as needed

Media switching

Redundant TDM switch

Distributed IP network (multiple path)

Intercabinet interfaces

Redundant

Distributed IP interfaces (multiple path)

Intracabinet buses

Redundant TDM bus

Redundant Ethernet buses

Power supplies

Redundant

Redundant

Line cards

Single (usually 24)

Not applicable

Power to phones

Inline (phantom)

Inline (phantom), third pair, or external

Phones

Single interface

Capable of registering with up to three CallManagers and one SRST for retention of service during network outages

CallManager redundancy works differently. The redundancy model differs by Cisco IP Communications component. Clustering has one meaning with regard to the database, another meaning with regard to CallManager nodes, and a third meaning with regard to the client devices.
Database Clustering

To serve calls for client devices, CallManager needs to retrieve settings for those devices. In addition, the database is the repository for information such as service parameters, features, and the route plan. The database layer is a set of dynamic link libraries (DLL) that provide a common access point for data insertion, retrieval, and modification of the database. The database itself is Microsoft SQL 2000.

If the database were to reside on a single machine, the phone network would be vulnerable to a machine or network outage. Therefore, the database uses a replication strategy to ensure that every server can access important provisioning information even if the network fails.

Each CallManager cluster consists of a set of networked databases. One database, the Publisher, provides read and write access for database administrators and for CallManager nodes themselves. For large installations, it is recommended that the Publisher reside on a separate server to prevent database updates from impacting the real-time processing that CallManager does as part of processing calls.

In normal operations, all CallManager nodes in a cluster retrieve information from the Publisher. However, the Publisher maintains a TCP connection to each node in the cluster that runs a CallManager. When database changes occur, the Publisher database replicates the changed information to Subscriber databases on each of these connected nodes. The Publisher replicates all information other than Publisher call detail records (CDR). In addition, the Publisher serves as a repository for CDRs written by all CallManager nodes in the cluster.

In a large campus deployment, a server is often dedicated to handling the Publisher database. This server is often a high-availability system with hardware redundancy, such as dual power supply and Redundant Array of Independent Disks (RAID) disk arrays.

Subscriber databases are read-only. CallManager nodes access the Subscriber databases only in cases when the Publisher is not available. Even so, CallManager nodes continue operating with almost no degradation. If the Publisher is not available, Subscriber nodes write CDRs locally and replicate them to the Publisher when it becomes available again. Figure 1-9 shows database clustering.
01fig09.gif

Figure 1-9 Database Clustering
CallManager Clustering

Although the database replicates nearly all information in a star topology (one Publisher, many Subscribers), CallManager nodes replicate a limited amount of information in a fully-meshed topology (every node publishes information to every other node).

CallManager uses a fully-meshed topology rather than a star topology because it needs to be able to respond dynamically and robustly to changes in the network. Database information changes relatively rarely, and the information in the database is static in nature. For example, the database allows you to specify which CallManager nodes can serve a particular device, but the information does not specifically indicate to which node a device is currently registered. Therefore, a star topology that prevents database updates but permits continued operation if the Publisher database is unreachable serves nicely.

CallManager, on the other hand, must respond to the dynamic information of where devices are currently registered. Furthermore, because processing speed is paramount to CallManager, it must store this dynamic information locally to minimize network activity. Should a node fail or the network have problems, a fully-meshed topology allows devices to locate and register with backup CallManager nodes. It also permits the surviving reachable CallManager nodes to update their routing information to extend calls to the devices at their new locations.

Figure 1-10 shows the connections between CallManager nodes in a cluster.
01fig10.gif

Figure 1-10 CallManager Clustering

When devices initialize, they register with a particular CallManager node. The CallManager node to which a device registers must get involved in calls to and from that device. Each device has an address, either a directory number or a route pattern. (See Chapter 2 for more information about call routing). The essence of the inter-CallManager replication is the advertisement of the addresses of newly registering devices from one CallManager to another. This advertisement of address information minimizes the amount of database administration required for a Cisco IP Communications network. Instead of having to provision specific ranges of directory numbers for trunks between particular CallManager nodes in the cluster, the cluster as a whole can automatically detect the addition of a new device and route calls accordingly.

The other type of communication between CallManager nodes in a cluster is not related to locating registered devices. Rather, it occurs when a device controlled by one CallManager node calls a device controlled by a different CallManager node. One CallManager node must signal the other to ring the destination device. The second type of communication is hard to define. For lack of a better term, it is called Intracluster Control Signaling (ICCS).

Understanding this messaging requires knowing more about CallManager architecture. CallManager is roughly divided into six layers:

* Link
* Protocol
* Aggregator
* Media Control
* Call Control
* Supplementary Service

Figure 1-11 depicts this architecture. At the beginning of each subsequent chapter of this book, there is a copy of this figure with shading to indicate the components of CallManager that are covered in that particular chapter.
01fig11.gif

Figure 1-11 Layers Within CallManager

The Link Layer is the most basic. Its function is to ensure that if a device sends a packet of information to CallManager, or CallManager sends a packet of information to a device, the sent packet is received. CallManager uses two methods of communication. The Transmission Control Protocol (TCP) is by far the most commonly used. TCP underlies much communication on the Internet. It provides for reliable communication between peers using the Internet Protocol. CallManager uses TCP for call signaling and media control with CallManager nodes, media devices, IP phones, H.323 gateways, and ISDN call signaling originating from MGCP gateways. The User Datagram Protocol is a protocol in which a sent packet is not guaranteed to be received. CallManager uses UDP for communication with MGCP gateways and SIP proxies. Although UDP itself is not reliable, MGCP or SIP is designed to handle instances where the IP network loses the message; in such a case, MGCP or SIP retransmit its last message.

The Protocol Layer includes the logic that CallManager uses to manage the different types of devices that it supports. These devices include media devices, trunk devices, and station devices. The Protocol Layer also supports third-party integration with CallManager through the TAPI and JTAPI protocols.

The Aggregator Layer allows CallManager to properly handle the interactions between groups of related devices. The media resource manager, for example, permits one CallManager node to locate available media devices, even if they are registered to other CallManager nodes. The route list performs a similar function for gateways. Line control permits CallManager to handle IP phones that share a line appearance, even if the IP phones are registered with different CallManager nodes.

The Media Control Layer handles the actual media connections between devices. It handles the media control portion of setting up a call, but it also handles more complicated tasks. For instance, sometimes CallManager must introduce a transcoding device to serve as an interpreter for two devices that don’t communicate via the same codec. In this case, one call between two devices consists of multiple media hops through the network. The Media Control Layer coordinates all the media connections.

The Call Control layer handles the basic call processing of the system. It locates the destination that a caller dials and coordinates the Media Control, Aggregator, and Protocol Layers. Furthermore, it provides the primitives that the Supplementary Service Layer uses to relate independent calls. The Supplementary Service Layer relates independent calls together as part of user-requested features such as transfer, conference, and call forwarding.

Within each layer, the SDL application engine manages state machines, which are essentially small event-driven processes, but they do not show up on the Microsoft Windows 2000 Task Manager. Rather, the SDL application engine manages state machine tasks. These state machines each handle a small bit of the responsibility of placing calls in a CallManager network. For example, one kind of state machine is responsible for handling station devices, whereas another type is responsible for handling individual calls on station devices.

These state machines perform work through the exchange of proprietary messages. Before CallManager release 3.0 was created, these messages were strictly internal to CallManager. With the 3.0 release, these messages could travel from a state machine in one CallManager node directly to another state machine managed by a different CallManager node. This mechanism is, in fact, what allows a CallManager cluster to operate with perfect feature transparency. The same signaling that occurs when a call is placed between two devices managed by the same CallManager node occurs when a call is placed between two devices managed by different CallManager nodes.

Architecturally, intracluster communication tends to occur at the architectural boundaries listed in Figure 1-11. Take, for example, the situation that occurs when two devices that share a line appearance register with different CallManager nodes. When someone dials the directory number of the line appearance, both devices ring. Even though the state machine responsible for managing each station is on its own CallManager node, both of these state machines are associated with a single state machine that is responsible for managing line appearances. (These can reside on one of the two CallManager nodes in question, or possibly on a third CallManager node.) The ICCS, however, guarantees that the feature operates the same, no matter how many CallManager nodes are handling a call.

The architectural layers are rather loosely coupled. In theory, a call between two devices registered to different CallManager nodes in the cluster could involve up to seven CallManager nodes, although in practice, only two are required.
Device Redundancy

In a traditional telephone system, the phone is a slave to the call processing logic in the cabinet; it is unaware of the operating condition of its master. Consequently, the secondary master must maintain the state of the endpoint. For this reason, traditional telephone system architectures are redundant architectures rather than distributed architectures: Maintaining state across more than a single backup processor is excessively complex and difficult. In the Cisco IP Communications architecture, the endpoint is aware of the operational status of the server, as well as its own connectivity states. As a result, the endpoints determine which CallManager nodes serve them. You can provision each endpoint with a list of candidate nodes. If the node to which an endpoint is registered has a software problem, or a network connectivity glitch prevents the endpoint from contacting the node, the endpoints move their registration to a secondary or even tertiary CallManager. Phones in active conversations, assuming that the media path is not interrupted, maintain their audio connection to the party to which they are streaming. However, because CallManager is not available to the phone during this interim, users cannot access features on the preserved call. When the call terminates and the phone reregisters, the phone regains access to CallManager features.

Figure 1-12 shows an example of this behavior in action. On Step 1 on the left, three phones are homed to CallManager SanJoseC in a cluster, and each has multiple CallManager nodes configured for redundancy. In Step 2, CallManager SanJoseC fails. As a result, Step 3 shows that all phones that were registered with CallManager SanJoseC switch over to their secondary CallManagers. One phone moves to CallManager SanJoseB, and the other phones move to CallManager SanJoseA.
01fig12.gif

Figure 1-12 Device Redundancy
Deployment of Servers Within a CallManager Cluster

Each CallManager node in a cluster can support up to 7500 phones. A CallManager cluster can support up to 30,000 phones. Adding multiple clusters permits as many phones as you need. Within a cluster, several strategies exist for deployment of servers. Servers can be arranged into clusters, built up of small “molecular” (for lack of a better word) units. Any individual cluster contains at most one Publisher database. Every non-Publisher server in the cluster contains a Subscriber database. Furthermore, each cluster must contain at least one TFTP server to provide Cisco IP Phones and gateways their configurations.

Often, a cluster needs to contain individual servers that run applications or media services (for annunciation or music on hold). From a call agent architectural standpoint, these servers are more akin to end devices than direct participants in the clustering model.

Any single cluster must be composed according to the following rules:

* A cluster can contain at most 20 servers.
* A cluster can contain at most eight nodes running CallManager.
* A cluster must have a Publisher database.
* A cluster must have at least one TFTP service running.

For survivability purposes, a given cluster must contain at least two CallManager nodes. In case one node fails, IP phones and gateways can fail over to the backup node for call processing services. Cisco recommends two models for call processing redundancy. You can compose a cluster using a combination of the two models, but, in general, only one of the two is employed.

In the 1:1 model, you can have either one node entirely in reserve or split the load evenly between the primary and secondary node. If the primary node should fail, all devices registered to it fail over to the secondary node. A 1:1 redundancy model allows you to support the maximum number of phones—7500—per individual node.

In the 2:1 model, you hold one node in reserve for every two nodes that host active devices. If either primary node should fail, the devices registered to that node fail over to the backup node. Because both primary nodes could, in theory fail, causing all devices on both primary nodes to rehome to the secondary node, any individual primary node cannot host the maximum number of devices without unduly stressing the secondary node should both primary nodes fail. Therefore, a 2:1 model allows you to support 5000 phones per primary node, yielding a total of 10,000 per 2:1 redundancy group.

The following sections describe and depict the different configurations.
Minimum Configuration—Up to 1250 Users

The minimum configuration consists of merely two servers. However, because these servers must host a primary CallManager, backup CallManager, Publisher and Subscriber databases, and TFTP server, the maximum number of Cisco IP Phones and gateways that can be supported is 1250.

In this model, one server houses the Publisher and Cisco TFTP, and it serves as a backup CallManager. The other server houses a primary CallManager. Under normal operating conditions, all devices in the cluster register to the second server, but if the second server is unavailable, the first server takes over CallManager responsibilities. Figure 1-13 shows this deployment model.
01fig13.gif

Figure 1-13 Deployment Model 1 for up to 1250 Users
1:1 Redundancy—Up to 7500 Users

To support more than 1250 users, you must dedicate at least one server to both a Publisher database and Cisco TFTP server. When the data management services are offloaded onto a separate server, you can dedicate servers specifically to call processing.

In a 1:1 redundancy group, one server acts as a primary call processing server, with a second server prepared to take over call processing services should the primary server fail. This model also permits load sharing—you can spread your users across both servers. Should a server fail, you can permit users served by the failed CallManager server to fail to the other server. Figure 1-14 shows this deployment model.
01fig14.gif

Figure 1-14 1:1 Redundancy Group with Separate Publisher and TFTP Server (7500 Users)
2:1 Redundancy—Up to 10,000 Users

To support more than 1250 users, you must dedicate at least one server to both a Publisher database and Cisco TFTP server. When the data management services are offloaded onto a separate server, you can dedicate servers specifically to call processing.

In a 2:1 redundancy group, two servers act as primary call processing nodes with one server reserved to take over call processing services should one or both primaries fail. The backup call processing node must be prepared to take the devices handled by both active call processing nodes. To prevent overloading the secondary nodes in case both primaries fail, the maximum number of devices that any primary can support must be reduced from 7500 to 5000, yielding a maximum load on the secondary of 10,000 devices should both primary nodes fail.

Figure 1-15 shows this deployment model.
01fig15.gif

Figure 1-15 2:1 Redundancy Model with Separate Publisher and TFTP Server (10,000 Users)

Separating the Publisher and TFTP server from the call processing nodes has the advantage of eliminating the risk that database activity on the Publisher node degrades performance of CallManager if the primary CallManager is unavailable.
Up to 30,000 Users

When you exceed 7500 users, it is advisable to configure one server as the Publisher database and one as a TFTP server.

After doing so, you can construct a cluster using either the 1:1 redundancy model or 2:1 redundancy model for call processing nodes.

The 1:1 redundancy model permits you to achieve the cluster maximum of 30,000 IP phones, with 1 server dedicated for a Publisher database, at least one server dedicated for TFTP, and four 1:1 redundancy groups. If you have 7500 IP phones per group, that yields 30,000 IP phones for each of 4 primary servers.

Figure 1-16 shows this deployment model.
01fig16.gif

Figure 1-16 Deployment Model for 15,000 to 30,000 Users
More than 30,000 Users

When the number of users climbs above 30,000, a single cluster cannot manage all devices. However, you can connect CallManager clusters together through either gateways or direct CallManager-to-CallManager connections called intercluster trunks. Intercluster trunks run a variant of the H.323 protocol. Figure 1-17 shows this configuration.

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Kevin Wallace, CCIE No. 7945

ISBN: 1-58705-321-7

As a final preparation tool providing a review of CIPT exam topics, the CCVP CIPT Quick Reference Sheets complement official Cisco curriculum, other books, or other exam preparatory material. This digital Short Cut provides readers with detailed, graphical-based information, highlighting the key topics on the latest CIPT exam in a quick-review format. These fact-filled Quick Reference Sheets allow certification candidates to get all-important information at a glance, helping them to focus their study on areas of weakness and to enhance memory retention of important concepts.

The CCVP certification recognizes a candidate’s ability to create an IP telephony solution that is transparent, scalable, and manageable. Earning a CCVP certification validates a robust set of skills in implementing, operating, configuring, and troubleshooting a converged IP network. The certification content focuses on Cisco Systems Unified CallManager, quality of service (QoS), gateways, gatekeepers, IP phones, voice applications, and utilities on Cisco routers and Cisco Catalyst switches.

The Cisco IP Telephony (CIPT) exam tests the candidate’s knowledge of voice-over-IP (VoIP) and public switched telephone network (PSTN) components and technologies and the candidate’s ability to describe, install, configure, and support Cisco CallManager version 4.1 in a Cisco network, including such features as security and video.

This Short Cut is derived from the print publication, Cisco IP Telephony Flash Cards and Exam Practice Pack, ISBN: 1-58720-128-3.

Table of Contents:

The Cisco CallManager

IP Telephony Components

Dial Plans

IP Telephony Options

IP Telephony Applications

Administrative Utilities

Cisco CallManager 4.x Enhancements

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Jeremy Cioara, CCIE® No. 11,727

Cisco IP Telephony (CIPT), Second Edition, is a Cisco®-authorized, self-paced learning tool for CCVP IP telephony foundation learning. This book provides you with the knowledge needed to install, configure, and maintain a Cisco IP telephony solution. By reading this book, you will gain a thorough understanding of Cisco Unified CallManager version 4.1, the call routing and signaling component for the Cisco IP telephony solution.

Cisco IP Telephony (CIPT) expands your knowledge of voice over IP (VoIP) and public switched telephone network (PSTN) components and technologies. This book teaches you how to describe, install, configure, and support Cisco Unified CallManager 4.1 in a Cisco network including such features as security and video; how to configure gateways, gatekeepers, and switches; and how to build route plans to place intra- and intercluster Cisco IP phone calls. Chapter review questions, detailed configuration examples, network diagrams, and thorough explanations help reinforce learning.

Whether you are preparing for CCVP certification or simply want to gain a better understanding of IP telephony and Cisco Unified CallManager, you will benefit from the foundation information presented in this book.

Cisco IP Telephony (CIPT), Second Edition, is part of a recommended learning path from Cisco Systems® that includes simulation and hands-on training from authorized Cisco Learning Partners and self-study products from Cisco Press. To find out more about instructor-led training, e-learning, and hands-on instruction offered by authorized Cisco Learning Partners worldwide, please visit www.cisco.com/go/authorizedtraining.

Jeremy D. Cioara, CCIE® No. 11,727, is the owner of AdTEC Networks and works as a network consultant, instructor, and author. He has been working in network technologies for more than a decade and has deployed networks worldwide. His current consulting work focuses on network and VoIP implementations.

* Examine design strategies behind a Cisco Unified CallManager cluster, cluster replication, and Cisco Unified CallManager deployment models
* Perform Cisco Unified CallManager server installations and upgrades
* Learn the features of all Cisco IP Phones, the IP phone startup process, and audio codec communication
* Add IP phone users and apply bulk moves, adds, and changes
* Configure Cisco gateways and trunks
* Design and configure Cisco Unified CallManager route plans
* Implement telephony call restrictions and control
* Effectively coordinate multisite deploymentsConfigure user features, Cisco Unified CallManager Attendant, and Cisco IP Manager Assistant
* Secure the Windows operating system and Cisco Unified CallManager administration
* Prevent toll fraud and harden the IP phone against attack
* Configure Cisco Unified CallManager to support video
* Monitor performance and configure alarms, traces, and CAR

This volume is in the Certification Self-Study Series offered by Cisco Press®. Books in this series provide officially developed self-study solutions to help networking professionals understand technology implementations and prepare for the Cisco Career Certifications examinations.

Category: IP Communications

Covers: Cisco Unified CallManager v4.1

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Understanding and configuring GW/GK in complex VoIP networks

Denise Donohue, CCIE® No. 9566

David Mallory, CCIE No. 1933

Ken Salhoff, CCIE No. 4915

Deployments of voice over IP (VoIP) networks continue at a rapid pace. Voice gateways are an essential part of VoIP networks, handling the many tasks involved in translating between transmission formats and protocols and acting as the interface between an IP telephony network and the PSTN or PBX. Gatekeepers and IP-to-IP gateways help these networks scale. Gatekeepers provide call admission control, call routing, address resolution, and bandwidth management between H.323 endpoints including Cisco IOS® voice gateways and Cisco® Unified CallManager clusters. IP-to-IP gateways allow VoIP calls to traverse disparate IP networks.

Cisco Voice Gateways and Gatekeepers provides detailed solutions to real-world problems encountered when implementing a VoIP network. This practical guide helps you understand Cisco gateways and gatekeepers and configure them properly. Gateway selection, design issues, feature configuration, and security and high-availability issues are all covered in depth. The abundant examples, screen shots, configuration snips, and case studies make this a truly practical and useful guide for anyone interested in the proper implementation of gateways and gatekeepers in a VoIP network. Emphasis is placed on the accepted best practices and common issues encountered in real-world deployments.

Cisco Voice Gateways and Gatekeepers is divided into four parts. Part I provides an overview of an IP voice network. Part II is dedicated to voice gateways, including discussions of Media Gateway Control Protocol (MGCP); H.323; Session Initiation Protocol (SIP); voice circuit options; connecting to the PSTN, PBX, and IP WAN; dial plans; digit manipulation; route selection; class of restriction; Survivable Remote Site Telephony (SRST) and MGCP fallback; digital signal processor (DSP) resources; and Tool Command Languaue (Tcl) scripts and Voice XML (VXML). Part III addresses voice gatekeepers, including detailed deployment and configuration. Part IV is dedicated to IP-to-IP gateways.

“With this book, the authors provide an in-depth look at the breadth of voice gateway features and capabilities, as well as providing voice gateway configuration guidance.”

–Christina Hattingh, Access Technology Group, Cisco Systems®

*
Understand the pros and cons of MGCP, H.323, and SIP, how they implement call signaling and describe call flow, and how to configure each protocol
* Learn the various analog and digital voice circuit options used to connect a VoIP network to the PSTN
* Configure and troubleshoot PSTN, PBX, and IP WAN connections
* Build scalable dial plans and understand the different types of dial peers
* Understand the various ways gateways control called and calling phone numbers
* Examine call admission control (CAC) techniques
* Configure Class of Restrictions (CoR) for both inbound and outbound calls
* Deploy and troubleshoot SRST and MGCP fallback
* Evaluate DSP considerations and resources
* Support interactive voice response (IVR) and advanced call-handing applications using Tcl scripts and VXML
* Deploy and configure basic and advanced gatekeeper functions
* Configure and troubleshoot IP-to-IP gateways

This book is also recommended self-study training for the CCVP GWGK exam.

This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press help networking professionals understand voice and IP telephony technologies, plan and design converged networks, and implement network solutions for increased productivity.

Category: Cisco Press–IP Communications

Covers: VoIP gateways and gatekeepers

$65.00 USA / $81.00 CAN

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