Professional Documents
Culture Documents
642-481 CRMC
The prerequisite for this Specialist certication is that the certication candidate must
currently be either CCNA certied or have any CCIE certication.
flast.indd xxi 9/20/11 10:56:04 AM
xxii Introduction
The exams necessary to achieve either of these two Cisco certications can be taken
in any order you choose, but it is very common to start with the 642-437 CVOICE v8.0
exam, because it provides a solid foundation for the remainder of the exams.
A Closer Look at Ciscos Voice Certifications
Probably most readers of this study guide will be looking to achieve their CCNP Voice
certication, because it is part of Ciscos core structure for voice. Cisco offers three
distinct levels of core voice certications. The following diagram shows that the CCNA Voice
certication is a building block to the professional- and expert-level voice certications:
CCIE
Voice
CCNP Voice
CCNA Voice
As of the writing of this book, the CVOICE v8.0 (642-437) exam costs $250 USD. The
exam tests your knowledge a great deal in areas both theoretical and technically specic to
Cisco hardware and software.
Once you use this book to pass the CVOICE v8.0 exam, you can choose to continue
on the CCNP Voice path and pass the other four exams to achieve the CCNP Voice
certication. If you choose to achieve the CCNP Voice certication, you may want to
further your education and attempt to pass the CCIE Voice certication. But even if
you stop after achieving your CCNP Voice certication, you will have demonstrated to
your current or prospective employers that you have professional-level knowledge of the
interoperations of legacy PSTN and Cisco voice technologies. This assurance to employers
will make it easier for you to land that dream job youve always wanted!
What Skills Do You Need to Pass the CVOICE v8.0 Exam?
To pass the 642-437 exam, you should be procient in the following areas:
The ability to install, configure, and support Cisco voice gateways and gatekeepers.
This includes functions including, but not limited to, dial peers, digit manipulation,
path selection, calling privileges, signaling protocols, DSP farms, and analog and
digital ports.
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Introduction xxiii
The ability to install, configure, and support a Cisco Unified Communications Manager
Express (CUCM Express) system and endpoints. This also includes preparation of
CUCM Express support components, including DHCP, NTP, and TFTP.
The ability to install, configure, and support a Cisco Unified Border Element (CUBE)
for functionality including address hiding, protocol/media internetworking, and call
admission control.
Chapter 2, Understanding Analog and Digital Voice, provides you with the
background covering traditional analog and digital telephony ports that are commonly
installed on voice gateways that connect to the PSTN or legacy PBX systems. Topics
such as network signaling, interface types, and the analog-to-digital conversion process
are covered in detail along with the basics of configuring many of these interfaces on
Cisco hardware.
Chapter 3, VoIP Operation and Protocols, introduces you to voice transport over
an IP network. Topics in this chapter include voice media transmission and control
protocols, voice gateway signaling protocols, and an introduction to common H.323
network components.
Chapter 4, The VoIP Path-Selection Process, provides you with the path-selection
process that a voice gateway goes through each time a call needs to be routed through
it. This includes a thorough understanding of the dial-plan selection process and
on- versus off-network calling. Additionally, we cover the differences between POTS
and VoIP dial peers and how to modify voice gateway path selections based on
dial-peer strategies and dial-peer wildcards, translations, and manipulation techniques.
Chapter 6, Configuring Voice Gateway Ports and DSPs, dives into more complex
voice gateway configuration techniques that show readers how to set gateway features
such as PLAR FXS/FXO DID, E&M bridge, and CAMA and explores several T1
scenarios. Additionally, you will learn how to configure a voice gateway as a DSP farm
for the offloading of services such as transcoding, conferencing, and MTP services.
Chapter 8, Configuring and Managing CUCM Express, introduces you to the world
of the CUCM Express router and the functionality it can provide small to medium-size
businesses and remote-site offices. Those preparing for the CVOICE v8.0 exam must
know not only how to configure the CUCM Express router but also how to prepare the
IP network for voice communications with CUCM Express. This includes topics such
as PoE for Cisco IP phones, voice VLAN configuration best practices, and network
services for the support of IP phones including DHCP, TFTP, and NTP.
Chapter 9, Advanced Voice Gateway Features, shows readers several of the value-
added features and functionalities of voice gateways. Some of these features help
to facilitate the exchange of calls between IP and legacy PSTN networks, as is the
case with DTMF and fax/modem relay. Youll also see how to configure fallback
functionality on networks that operate both IP and PSTN networks between sites.
Lastly, we take a look at some cost-saving features inherent when you configure
features such as TEHO and call blocking.
Multiple-choice single-answer
Drag-and-drop
flast.indd xxvii 9/20/11 10:56:07 AM
xxviii Introduction
Fill-in-the-blank
Arrive at least 30 minutes early to the exam center. That way you can check in and
mentally prepare for the exam without having to rush.
Take the Cisco exam tutorial found within the Cisco exam software on test day. This
tutorial is offered prior to the official start of each exam before the test timer starts.
In this tutorial you will be given an interactive lesson as to the format of the exam
and how to navigate through the different question types, including multiple-choice,
drag-and-drop, fill-in-the-blank, and simulation questions. Even if you have taken
many Cisco exams, I highly recommend going through the tutorial in case there is
something new to the exam format since the last time you took an exam.
Read both the questions and answers very carefully. Cisco often will intentionally lead
the hasty test taker, who simply glosses over a question, to quickly choose the incorrect
answer. Patience and careful thinking pay off greatly when taking Cisco exams!
Be aware that you cannot go back to change an answer once you have moved on to the
next question. Make sure that the answer you choose is the one you want to stick with,
because there is no way to change it later on.
Conventions Used in This Book
This book uses certain typographic styles in order to help you quickly identify important
information and to avoid confusion over the meaning of words such as on-screen prompts.
In particular, look for the following styles:
Italicized text indicates key terms that are described at length for the first time in a
chapter and are defined in the books Glossary.
Bold monospaced text is information that youre to type into the computer, usually at a
command prompt.
In addition to these text conventions, which can apply to individual words or entire
paragraphs, a few conventions highlight segments of text:
A note indicates information thats useful or interesting but thats somewhat
peripheral to the main text. A note might be relevant to a small number of
networks, for instance, or it may refer to an outdated feature.
A tip provides information that can save you time or frustration and that
may not be entirely obvious. A tip might describe how to get around a
limitation or how to use a feature to perform an unusual task.
Warnings describe potential pitfalls or dangers. If you fail to heed a
warning, you may end up spending a lot of time recovering from a bug, or
you may even end up restoring your entire system from scratch.
Sidebars
A sidebar is like a note but longer. The information in a sidebar is useful, but it doesnt
t into the main ow of the text.
Real World Scenario
A real world scenario is a type of sidebar that describes a task or example thats
particularly grounded in the real world. This may be a situation that I or somebody I
know has encountered, or it may be advice on how to work around problems that are
common in real, working Cisco environments.
How to Contact Sybex
Sybex strives to keep you supplied with the latest tools and information you need for your
work. Please check our website at www.sybex.com/go/cvoice, where well post additional
content and updates that supplement this book should the need arise.
flast.indd xxix 9/20/11 10:56:08 AM
Assessment Test
1. After the AutoQoS for the Enterprise implementation phase has been completed, what final
step should be done?
A. Disable the discovery phase process within every interface it is running by issuing the
no auto discovery qos command.
B. Disable the discovery phase process globally by issuing the no auto discovery qos
command.
C. Schedule the autodiscovery phase process to run every week within every interface by
issuing the auto discovery qos 7 command.
D. Schedule the autodiscovery phase process to run every week globally by issuing the
auto discovery qos 7 command.
2. Which of the following DTMF relay methods transmit tones in an ASCII format? (Choose
all that apply.)
A. h245-signal
B. h245-alphanumeric
C. cisco-rtp
D. rtp-nte
3. Given the following information, what UC deployment model should you choose if your
business has six large (1,000 users or more) and geographically dispersed campuses that are
interconnected together by a 3 Mbps WAN link?
A. Centralized services model
B. Distributed services model
C. Inter-networking of services model
D. Geographical diversity model
4. Which of the following commands is the correct syntax and interface mode to configure
AutoQoS for VoIP on a Cisco router?
A. Router(config-if)#auto qos voip
B. Router(config-if)#auto qos voip cisco-phone
C. Router(config)#auto qos voip
D. Router(config)#auto qos voip cisco-phone
5. What is the correct command used to configure loop-start signaling on an FXS port?
A. Router(config-voiceport)#dial-type loopstart
B. Router(config-controller)#dial-type loopstart
C. Router(config-controller)#signal loopstart
D. Router(config-voiceport)#signal loopstart
flast.indd xxx 9/20/11 10:56:09 AM
6. Which of the following is not a feature of a Cisco Unified Border Element (CUBE)?
A. Call admission control (CAC)
B. Secure deployment
C. IP address hiding
D. Zone management
7. Dr. Nyquist discovered that analog samples taken at times the highest frequency would
produce high-quality sound when reconstructed using only the taken samples.
A. Three
B. Two
C. Five
D. Four
8. Which of the following is not a voice gateway signaling protocol?
A. MGCP
B. SCCP
C. Q.931
D. H.323
9. What type of voice trunk directly connects a private switch to a public switch?
A. CO trunk
B. Interoffice trunk
C. Tie trunk
D. Tandem trunk
10. What H.323 device maintains a database of telephone extensions to IP address mappings?
A. Proxy server
B. MCU
C. Gateway
D. Gatekeeper
11. A phone call enters a voice gateway. What happens if no incoming dial peer is matched?
A. The call is routed out the PSTN by default.
B. The call is dropped.
C. The voice gateway sends a redirect signal to the calling phone.
D. The call will match the default dial peer.
Assessment Test xxxi
flast.indd xxxi 9/20/11 10:56:10 AM
12. How are the voice and native data VLANs treated differently on the link between the Cisco
switch and the Cisco IP phone?
A. The voice VLAN is tagged using 802.1Q and the data VLAN is not tagged.
B. The voice VLAN is tagged using ISL and the data VLAN is tagged using 802.1Q.
C. The voice VLAN is not tagged and the data VLAN is tagged using ISL.
D. The voice VLAN is not tagged and the data VLAN is tagged using 802.1Q.
13. The following destination pattern is configured in a dial peer:
Router(config-dial-peer)# destination-pattern 34.?
Which of the following dial strings will be matched? (Choose all that apply.)
A. 3484
B. 34
C. 342
D. 3433
14. According to the ITU-T G.114 specification, packet delay for voice should not
exceed ms.
A. 30
B. 50
C. 150
D. 250
15. What is the correct configuration command for setting a voice gateway to use ISDN switch
type primary-5ess?
A. Router(config)#isdn switch-type primary-5ess
B. Router(config-controller)# isdn signaling switch-type primary-5ess
C. Router(config-controller)# isdn switch-type primary-5ess
D. Router(config)# isdn signaling switch-type primary-5ess
16. What voice gateway feature replaces lost packets with ones that are intelligently generated?
A. PESQ
B. DSP
C. PLC
D. iSAC
17. What codec quality tool has been developed to better test and grade next-generation codecs
that use wideband?
A. MOS
B. POLQA
C. PSQM
D. PESQ
xxxii Assessment Test
flast.indd xxxii 9/20/11 10:56:10 AM
18. Which of the following are limitations inherent in loop-start signaling? (Choose all that apply.)
A. It is unable to properly transition on-hook for inbound calls when FXO interfaces are used.
B. Glare.
C. It is unable to properly transition off-hook for inbound calls when FXO interfaces are used.
D. Gleam.
E. It is unable to properly transition on- or off-hook for inbound calls when FXO
interfaces are used.
19. What FXS config-voiceport command can be used to adjust the analog ring tone?
A. ring frequency
B. ring cadence
C. ring type
D. cptone
20. How many simultaneous calls can an E1 CAS circuit support?
A. 24
B. 31
C. 32
D. 30
21. Which of the following commands can be used to verify the line coding of a T1 interface?
A. show voice port
B. show voice port summary
C. show controller t1
D. show interface
22. When an H.323 gatekeeper receives an ARQ message from a registered H.323 device, what
two decisions does the gatekeeper make about a requested call?
A. What codec should be used
B. What type of H.323 device is attempting to make the call
C. Whether the call is permitted to go through
D. How the call should be routed
23. What voice signaling protocol is used by default when configuring dial peers on a router
with an IP voice gateway IOS?
A. SIP
B. SCCP
C. H.323
D. MGCP
E. SIPv2
Assessment Test xxxiii
flast.indd xxxiii 9/20/11 10:56:10 AM
24. Which of the following best describes an on-net to off-net call?
A. An internal user calling a telephone accessed through the PSTN
B. An internal user calling a remote site through the secondary PSTN path during a
WAN failure
C. An external user calling a remote site through the secondary PSTN path during a
WAN failure
D. An internal user calling a telephone accessed through the IP WAN
25. Which of the following is a common reason for adjusting the maximum number of SIP
retries?
A. If a high-compression codec is being used.
B. If the network is unreliable.
C. If the SIP gateway connects to an ISDN circuit.
D. If the SIP gateway accepts both TCP and UDP messages.
26. You are reviewing a routers configuration and see the following:
ephone 1
mac-address 0033.1c43.2533
type 7965
codec g729r8
button 1:1
What does the codec g729r8 command mean?
A. This ephone will operate only with the codec specified.
B. This is the preferred codec for the ephone.
C. This is the only codec that the IP phone understands.
D. DSP resources have been specifically set aside for this ephone.
27. Voice packets reach the destination IP phone with a delay variance between 15 and 50 ms.
What is the result?
A. The packets will be dropped.
B. Queuing buffers in the phone will smooth out any jitter.
C. The destination phone will reject the call by sending back a reorder signal to the
calling party.
D. The stream may sound garbled because it exceeds best-practice limits.
xxxiv Assessment Test
flast.indd xxxiv 9/20/11 10:56:11 AM
28. When viewing show ephone output like the following, what does SEIZE mean on
the extension?
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:1 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 16 max_line 6
button 1: dn 2 number 4002 CH1 SIEZE
Preferred Codec: g711ulaw
Active Call on DN 2 chan 1 :4002 0.0.0.0 0 to 0.0.0.0 0 via 0.0.0.0
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn -1
A. The phone is currently in a call.
B. The phone is on-hook.
C. The phone is off-hook and unregistered.
D. The phone is off-hook.
E. The phone is receiving a call.
29. CAMA interfaces physically connect to what destination?
A. A PBX
B. The PSTN
C. The PSAP
D. A DID
30. Which of the following correctly configures a call-block profile (called block_976) for
incoming calls on a POTS dial peer?
A. call-block translation-profile block_976 incoming
B. call-block translation-profile incoming block_976
C. translation-profile call-block incoming block_976
D. translation-profile call-block block_976 incoming
31. What two methods are used to transmit RAS location messages?
A. Round-robin
B. Sequential
C. FIFO
D. Blast
Assessment Test xxxv
flast.indd xxxv 9/20/11 10:56:11 AM
32. Which of the following QoS variable-delay reduction techniques might use CBWFQ?
A. Prioritize time-sensitive traffic
B. Link fragmentation and interleaving
C. Compression
D. Bandwidth upgrades to eliminate bottlenecks
33. What markings can Cisco Catalyst L2 switches use to enforce QoS?
A. DSCP
B. IP Precedence
C. CoS
D. RSVP
34. When configuring MQC, what command is used to associate traffic class types with one or
more QoS operations?
A. class-map
B. policy-map
C. traffic-map
D. qos-map
35. Which telephony edge device converts voice into a binary stream?
A. PBX
B. Digital telephone
C. CO trunk
D. Tie trunk
36. What must be carefully watched when cRTP is configured between two voice gateways?
A. Packet fragmentation
B. Gateway CPU utilization
C. Packet delay
D. Packet jitter
37. What is the process of translating between two different codecs?
A. Transcoding
B. MTP
C. Translation
D. DSP
38. What is the proper name for the international numbering plan that was developed by the ITU?
A. G.711
B. NANP
C. E.164
D. E.711
xxxvi Assessment Test
flast.indd xxxvi 9/20/11 10:56:12 AM
Answers to Assessment Test xxxvii
Answers to Assessment Test
1. A. As soon as you have implemented AutoQoS for the Enterprise policies, you no longer
need to waste CPU resources by keeping the discovery phase running on an interface.
To disable the autodiscovery process, you should go into interface conguration mode of
each interface the processes is running and issue the no auto discovery qos command.
See Chapter 12.
2. A, B. Both the h245-alphanumeric and h245-signal DTMF relay methods convert tones
to ASCII for transmission on IP networks. See Chapter 9.
3. C. The best choice would be the inter-networking of services model because of the distributed
nature of the multisite network and the fact that the WAN links (3 Mbps) are likely to be too
small to transport voice trafc to a centralized call-processing agent. See Chapter 1.
4. A. The correct syntax for AutoQoS for VoIP on a router is auto qos voip. This command
is performed while in interface conguration mode. See Chapter 12.
5. D. Because an FXS port is an analog connection, you will be in config-voiceport mode.
The correct command while in this mode is signal loopstart. See Chapter 2.
6. D. Zone management is a feature of an H.323 gatekeeper and not a CUBE. See Chapter 10.
7. B. Sampling at a rate of twice the highest frequency to be represented follows the Nyquist
sampling theorem. See Chapter 2.
8. C. Q.931 is not a voice gateway signaling protocol. See Chapter 3.
9. A. A CO trunk is the name used to describe a circuit that connects a private PBX switch to
a public switch at the central ofce. See Chapter 1.
10. D. An H.323 gatekeeper is a server that maintains a database of telephone extensions to
IP address mappings. Before a call is made, the gatekeeper must be queried to identify the
location of the destination H.323 endpoint. See Chapter 3.
11. D. If a call is not matched against congured incoming dial peers, it is matched against the
default dial peer (dial peer 0) and processed accordingly. See Chapter 4.
12. A. Voice VLANs are tagged with 802.1Q and the native data VLAN is left untagged. See
Chapter 8.
13. B, C. The . means that any digit can be used. The ? means that the previous digit or group
will occur 0 or one time. That means that 34 and 342 will be the two choices that match
this destination pattern. See Chapter 4.
14. C. The ITU-T recommends that end-to-end delay should not exceed 150 milliseconds for
voice packets. See Chapter 5.
15. A. The ISDN switch type is congured globally in cong mode. The correct command is isdn
switch-type primary-5ess in order to set the ISDN switch the voice gateway will connect to.
See Chapter 2.
flast.indd xxxvii 9/20/11 10:56:12 AM
xxxviii Answers to Assessment Test
16. C. Packet loss concealment is a software process that replaces lost packets with ones
intelligently derived by the router. See Chapter 5.
17. B. The Perceptual Objective Listening Quality Analysis tool is an ITU-T standard that is
being developed to test and score high-delity codecs. See Chapter 5.
18. B, C. Glare can be a big problem for telephone loop-start users who make and receive
frequent telephone calls. Also, there is not a proper way for FXO ports to properly go
off-hook at the end of a call that came inbound on the interface. See Chapter 2.
19. B. The ring cadence command is used to adjust the ring tone. See Chapter 6.
20. D. Although it uses robbed-bit signaling, an E1 CAS circuit uses 2 of its 32 channels for
framing and synchronization. Therefore it can support up to 30 simultaneous calls. See
Chapter 2.
21. C. The show controller t1 command displays conguration information for T1 and E1
ports. See Chapter 6.
22. C, D. When an H.323 device attempts to make a call that utilizes an H.323 gatekeeper,
that call request goes to the gatekeeper. The gatekeeper rst determines if the call is
permitted and then uses the E.164 destination address to determine what IP address the call
should be routed to. See Chapter 10.
23. C. H.323 is the default voice gateway signaling protocol. If you want to use a different
signaling protocol, you must manually specify it. See Chapter 7.
24. B. On-net to off-net calls occur when a call is made to a remote site but for some reason
the call cannot be completed on the IP WAN. A secondary path is used to establish the call
instead using the PSTN network. See Chapter 4.
25. B. If your network is prone to packet drops and/or congestion, it is common to raise the
maximum number of SIP retry messages to help ensure that SIP messages are properly
received between endpoints. See Chapter 7.
26. B. The codec command species the preferred codec for an ephone when this phone is
calling another phone that is also congured on CUCM Express. The command can be
used while conguring individual ephones. See Chapter 8.
27. D. The maximum jitter is 30 ms between voice packets. Because this call exceeds those
limits, the result may be a voice stream that sounds garbled at the destination phone. See
Chapter 11.
28. D. When a user picks up the phone handset, the phone goes into an off-hook state. This is
referred to as a line seizure. See Chapter 8.
29. C. CAMA interfaces are used to connect to the PSAP for E911 calling. See Chapter 6.
30. B. The proper syntax is call-block translation-profile incoming block_976. This
command is performed while in config-dial-peer conguration mode. See Chapter 9.
flast.indd xxxviii 9/20/11 10:56:12 AM
Answers to Assessment Test xxxix
31. B, D. The sequential method sends an LRQ to remote gatekeepers one at a time and waits
for a response before sending another message. The blast method sends LRQ messages to
all remote gatekeepers at one time. See Chapter 10.
32. A. Trafc prioritization techniques can use CBWFQ as a way to segment trafc on a
network and give one class higher priority over another. See Chapter 11.
33. C. Cisco Layer 2 switches can read and enforce QoS using CoS markings found in Ethernet
frames. See Chapter 11.
34. B. The policy-map command associates trafc classes (segmented using class maps) and
applies QoS operations to them. See Chapter 12.
35. B. The two types of traditional telephony edge devices are analog and digital telephones.
Digital telephones take an analog stream and digitize it for transport. See Chapter 1.
36. B. cRTP is very CPU intensive and can cause the CPU to spike, which can end up causing
packet drops. See Chapter 3.
37. A. Transcoding is the process of translating between two codecs. DSP resources are used to
ofoad transcoding. See Chapter 5.
38. C. The ITU International numbering plan is formally known as E.164. See Chapter 4.
flast.indd xxxix 9/20/11 10:56:13 AM
flast.indd xl 9/20/11 10:56:13 AM
An Introduction to
Traditional Telephony
and Cisco Unified
Communications
THE FOLLOWING CVOICE EXAM
OBJECTIVES ARE COVERED IN THIS
CHAPTER:
Describe the components of a gateway.
Chapter
1
c01.indd 1 9/21/11 12:09:10 PM
Evolution is the process of something changing over time
into a more complex state where it can better adapt to its
environment. Evolution typically is triggered only when
outside forces require changes to be made. Technology also evolves into newer and more
useful tools over time. While the analog phone is still around, advances have been made
and telephones have evolved into fully digital devices. Even more recently, weve seen
more and more voice running over IP networks that share the same cabling and routing
functions with data networks.
But throughout this telephone evolution process, many of the traditional interfaces,
signaling protocols, and setups remain unchanged. In order to understand voice networks
of today, we must rst take a step back in time to discuss traditional telephony topics. Once
you have a solid foundation, you can see how many of these elements have either remained
the same or evolved over time to improve voice networks as they transition from circuit-
switched networks to packet-switched networks.
Chapter 1 will start off covering traditional telephony devices. This includes legacy
analog and digital phones as well as a look at components within public telephone networks.
We will then move on to the two private telephone network types in most organizations.
Lastly, this chapter will cover Ciscos take on IP telephony networks and how it breaks down
components into separate functionality categories and deployment models.
Understanding Traditional Telephony
Components
In 1875, Alexander Graham Bell invented the telephone, a device that transmits and
receives sound, most commonly human speech. The telephone houses a microphone that
callers speak into. With a standard analog telephone, the speech is then transported across
a pair of copper wires in the form of an electrical signal.
As the popularity of telephones grew, companies began providing a telephone network
that was used to interconnect multiple phones throughout a region. Today, public telephone
networks are a mixture of analog and digital circuits and trunks that interconnect and
cover the globe.
Telephone systems can be split into public and private sections. The private side consists
of equipment owned and maintained by an individual user or business. The public side
is owned and maintained by the telephone company, and this service is paid for by the
c01.indd 2 9/21/11 12:09:11 PM
Understanding Traditional Telephony Components 3
individual or business owner who wants to use public phone services. The public switched
telephone network (PSTN) is the network that interconnects telephones found in homes
and businesses throughout towns, cities, countries around the world. It used to be that the
PSTN consisted solely of analog circuits. The rst analog circuit was just two wires, and it
was responsible for carrying a single telephone call. As technology improved, both the
public- and private-side equipment became more sophisticated. Private businesses could
own and maintain their own phone switches. These phone switches could then be
interconnected by trunk lines that were specically designed for the transport of voice
services between phone switches. In this rst section, we will investigate the traditional
telephony components that make up the private and public telephone network.
Telephony Edge Devices
The edge is the part of the phone system that end users interact with to make and receive
calls in their purest form. Traditional telephony edge devices can be divided into two
categories: analog and digital telephones. But even traditional telephony devices have
evolved to include more advanced features to make the calling experience a better one.
Here is a closer look at each of these phone types.
Analog Telephones
Analog edge devices are still somewhat common in homes and small business
environments. The analog telephone is commonly directly connected to the PSTN, so
all of the backend intelligence is the responsibility of the service provider, and the phone
user is simply responsible for purchasing and maintaining their analog telephone, which
is a very simple device. Some businesses still use analog PBX (private branch exchange)
systems, although they are becoming rare. Connecting an analog phone to a PBX provides
additional capabilities to the phone such as voicemail with message-waiting indicators, call
hold, and personalized ringtones. Other than that, the features of analog telephones are
very limited.
Digital Telephones
Digital telephone devices use special hardware to convert analog voice streams into a
digital data stream. Most legacy PBX systems are digital. It is also important to note that
the digital handsets of most of these digital PBX systems are proprietary. It is rare to be
able to mix and match different digital phones within a single digital PBX.
Phone Switches
On the public side of the overall telephony, there are public phone switches and private phone
switches. A PBX or key system can be installed by a private party to provide a multitude
of private telephone services to phones located within this private network. The differences
between a PBX and a key system are detailed later in this chapter. Extension-to-extension
c01.indd 3 9/21/11 12:09:12 PM
4 Chapter 1
o
w
FI GURE 1. 6 The PSTN local-to-international hierarchy
c01.indd 9 9/21/11 12:09:15 PM
10 Chapter 1
Chapter
2
c02.indd 33 9/21/11 11:47:48 AM
Unied Communications today still relies heavily on the
ability to connect to both legacy PBX systems and the public
telephone network. While it would be great to have an
end-to-end IP voice solution for every situation, that is simply not possible in many
businesses today. In reality, you will probably need to support legacy analog or digital
endpoints and circuits at some point.
This chapter provides a thorough introduction to analog and digital voice ports and
signaling protocols. We will also cover the analog-to-digital conversion process needed to
transform analog waveforms into binary code for transport on digital circuits. Once weve
covered the details of analog and digital telephony, you will learn how to congure the
basic settings on various analog and digital ports that are available on Cisco voice gateway
hardware.
Understanding Analog Voice
Ports and Signaling
Analog voice was the method used by the very rst telephones. The technology captures
sound and places it onto the wire using electrical currents. The process is fairly simple and
has worked now for 130 years or so, ever since the telephone was invented. While analog
ports are becoming extinct, there still are a number of situations where youre likely to
encounter analog devices and analog ports in your career. The following section covers
analog voice ports and their signaling techniques.
Analog Voice Port Types
From a Cisco perspective, there are three analog ports that you need to become familiar
with: FXS, FXO, and E&M. While many more analog port types are available out in the
wild, these are the three port types available as modules on Cisco voice gateway hardware.
Foreign Exchange Station Ports
Foreign Exchange Station (FXS) ports are used to connect plain old telephone service
(POTS) end devices to a voice gateway. FXS ports are also found in homes that directly
connect to the PSTN. FXS ports use two-wire cabling with RJ11 connectors.
c02.indd 34 9/21/11 11:47:49 AM
Foreign Exchange Office Ports
Foreign Exchange Ofce (FXO) ports connect the PSTN to a voice gateway. FXO ports
use the same two-wire RJ11 cabling that FXS ports utilize.
E&M Ports
E&M ports interconnect two PBX systems. The cabling uses either six or eight wires, which
are bundled into pairs of two. Unlike FXS and FXO ports, E&M ports can either use one
pair (two-wire) or two pairs (four-wire) for signaling purposes. This leaves two pairs for the
transport of voice communication. If you are not familiar with RJ48 cabling, it uses the same
eight-position, eight-contact (8P8C) modular connector that Ethernet uses. The difference
between RJ48 and RJ45 is in how the pins are wired. See E&M Signaling later in the
chapter for more about this signaling type.
Analog Voice Signaling
One of the rst technical objectives CVOICE candidates need to understand is how analog
telephones work on the PSTN. Telephones and telephone switches use signaling methods to
communicate various stages in the setup, transport, and teardown of a telephone call. Three
analog signaling categories will be covered in this section. Briey, they can be described as
the following:
Address Signaling Address signaling is the transmission of telephone digits from the
calling party phone to the called party phone. A unique sequence of digits identies each
individual phone on the network so the call reaches the correct destination.
Informational Signaling Informational signaling is feedback generated from the telephone
switch to the user in the form of tones or voice messages to inform the phone user what
state a call is in.
Supervisory Signaling Supervisory signaling detects changes in the status of the telephone
physical loop or trunk. The signaling is then used to set up and tear down calls. Loop-start
and ground-start analog signaling fall within this signaling category.
In addition to these three analog signaling categories that deal with signaling from the
customer premise equipment (CPE) to the PSTN, a separate set of signaling categories will
be detailed that cover signaling specically for E&M ports.
Address Signaling
A telephone number consists of a string of digits that uniquely identies a specic telephone
or telephone system on a voice network. When someone wishes to call another user, they pick
up a telephone handset and dial the unique digits that specify the telephone of the person they
wish to talk to. The interpretation and handling of the dialed digits are the responsibility of
address signaling. Two main methods are used to input telephone numbers using a telephone:
Pulse dialing
DTMF dialing
Understanding Analog Voice Ports and Signaling 35
c02.indd 35 9/21/11 11:47:49 AM
36 Chapter 2
Describe H.323.
Describe MGCP.
Describe SIP.
Chapter
3
c03.indd 77 9/21/11 11:16:21 AM
The last chapter covered the process of taking an analog signal
and processing it so it can be transported over digital circuits.
This process gets us one step closer to Voice over IP. Because
voice packets are already in a digital format, all we have to do is wrap the voice payload in an
IP packet, and it is ready for transport on an IP network. That is the rst topic of discussion
for Chapter 3. This process of packetizing voice signals for transport over an IP packet is
accomplished using RTP and RTCP. In addition, there are extensions to RTP that can be
used to decrease the header size of an IP voice packet and to transport the payload in a secure
manner. Well discuss these extensions, cRTP and sRTP, in detail, and youll see how and
when they can be used to improve call quality and secure transmissions.
Next, we will cover the four voice gateway signaling protocols: SIP, MGCP, SCCP, and
H.323. That discussion will also include an introduction to H.323 gatekeeper hardware and
common components specically found in H.323 networks. Lastly, we will cover various
situations in which one gateway signaling protocol would be preferred over another.
Voice Media Transmission Protocols
When you have a voice sample that has been converted to a digital format, you need to
include additional information so the voice payload can be sent to the intended destination
over an IP network. The information needed includes details such as the source and
destination IP address and transmission protocol and port used. Also, real-time trafc such
as voice requires additional protocol assistance for proper transport to a destination over IP.
The primary two protocols that accomplish this goal are RTP and its helper protocol, RTCP.
In addition, there are certain situations where the information stored within an RTP
packet header can be reduced so it can be more efciently sent over low-speed serial
connections. This is an extension of RTP called cRTP. Finally, well discuss how to
congure voice gateways to provide for secure transport of IP voice packets using sRTP.
Introduction to the Real-Time Transport Protocol
The Real-time Transport Protocol (RTP) was originally dened in IETF RFC 1889
and revised to its current standard, which is RFC 3550. The protocol was developed to
transport streaming data. By streaming data, we are specically talking about
real-time transport of voice and video. Because real-time transport of streaming data
occurs instantly, lost or damaged packets have no need to be resent. If the packets were
c03.indd 78 9/21/11 11:16:22 AM
Voice Media Transmission Protocols 79
resent, they would arrive at their destination late and/or out of order, and would be essentially
useless by the time the packet arrived. Therefore, RTP was designed to be used with the User
Datagram Protocol (UDP) instead of the Transmission Control Protocol (TCP).
UDP is a transport mechanism for IP packets that, unlike TCP, does not attempt to
retransmit or reorder packets that never arrive or are late to the destination. For this reason
and because UDP packets, lacking these features, are smaller than in TCP, UDP is an
ideal Layer 4 transport mechanism for both voice and video. UDP also offers multiplexing
capabilities for easy replication using multicasting protocols at upper layers of the OSI
model. In addition, UDP provides error-detection mechanisms that help make it both fast
and efcient on an IP network.
RTP functions strictly as an end-to-end protocol. This means that the IP source and
destination devices communicate RTP directly with each other, unlike those voice signaling
protocols that communicate with intermediary systems. For example, Figure 3.1 shows a
small network with two IP phones attached to it. They are using the Cisco proprietary SCCP
signaling protocol. When IP-phoneA wants to call IP-phoneB, the phone communicates to
the Cisco call processing agent (a CUCM). The CUCM then nds the location of IP-phoneB
and is responsible for the call setup. But as soon as the CUCM has established an end-to-end
call, the actual transport of voice packets goes directly between endpoints.
FI GURE 3.1 RTP end-to-end transport
M
CUCM
Switch
RTP packet ow
IP-phoneB IP-phoneA
S
C
C
P
s
i
g
n
a
l
i
n
g
S
C
C
P
s
i
g
n
a
l
i
n
g
The RFC for RTP does not specify the actual UDP ports that RTP should utilize. The
one requirement stated is that the UDP port must be an even number. Most voice networks
are set to use default RTP settings, which use random even-numbered UDP ports in the
range of 16384 to 32767 for the purpose of RTP transport. The RFC species that RTP
must always use even-numbered ports while RTCP uses odd-numbered ports. When a
connection is made between IP voice endpoints such as two IP phones, an even-numbered
UDP port is selected for the RTP packets to use from the source IP to the destination IP.
c03.indd 79 9/21/11 11:16:23 AM
80 Chapter 3
Configure dial-peers.
Chapter
4
c04.indd 103 9/21/11 11:17:03 AM
In this chapter we begin to explore just how it is that a voice
gateway makes call-routing decisions. When a call enters
a voice gateway, a router must have the intelligence to use
information such as source and destination telephone extensions to route the call properly
out of the voice gateway to the proper destination. In addition, these telephone numbers
may need to be modied on the voice gateway before forwarding the call setup information
to the next destination on a voice network.
This chapter will cover what dial plans and dial peers are and how to congure them.
Well then move on to examine the difference between dial peers and call legs and explore
the digit-manipulation techniques used on voice gateways.
Understanding the Dial Plan
Path-Selection Process
Dial plans are congured on voice gateways using dial peers to determine how calls are
directed through the IP and PSTN networks. In addition to path-selection responsibilities,
dial plans provide the following primary tasks:
Digit Manipulation The modication of dialed digits prior to routing a call out of the
voice gateway
Calling Privileges The permission or denial of a caller to certain destinations
This section will rst cover the different call types all voice calls can be categorized
under. Next, we will examine call routing and path-selection techniques and the process
of matching dial peers. Finally, we will look at path-selection strategies that can be used to
streamline dial plans for ease of use and cost-savings benets.
Understanding Voice Call Types
Voice calls are categorized into call types based on the location of the source and destination
phones relative to the IP and PSTN networks. Depending on the type of call being made,
dial plans must be congured differently to ensure optimal paths at the lowest cost. In
the following diagrams of voice call types, you will see the portion of the end-to-end calls
designated within a circle. Anything outside the circle is handled by the PSTN and other
voice networks outside of managerial control.
c04.indd 104 9/21/11 11:17:05 AM
Understanding the Dial Plan Path-Selection Process 105
Local Calls
When the source and destination phones are connected to the same call processing agent
or voice gateway, it is considered to be a local call. Figure 4.1 shows an example of an IP
phone calling an analog phone. Both of these endpoints use the same voice gateway, so the
call is considered to be local.
FI GURE 4.1 A local call
IP phone Analog phone
V
Switch
Gi1/0 FXS0/0/0
Voice GW
On-Net Calls
When the source and destination phones are on the same network but traverse more than
one voice gateway, it is considered to be an on-network or on-net call. Because the call
is carried over a private network as opposed to the PSTN, there is no per-minute cost
incurred. Figure 4.2 shows the path between two IP phones located at different locations
but interconnected through an IP WAN. A call made between these two phones must be
processed by two voice gateways.
Switch
IP phone IP phone
Switch
V V
S0/1 S0/1 Gi1/0 Gi1/0
Voice GW Voice GW
IP WAN
FI GURE 4. 2 An on-net call
c04.indd 105 9/21/11 11:17:05 AM
106 Chapter 4
Chapter
5
C05.indd 145 9/21/11 11:18:46 AM
When you begin the design process for a VoIP network, there
are many decisions that need to be made prior to implementa-
tion. First, you need to understand the full capabilities of voice
gateway DSPs and how they can be used to ofoad processor-intensive tasks from the call-
processing agent. To design a network you must also understand some unique factors found
in VoIP networks, including VAD and network-related issues such as latency, jitter, and
packet loss. You can then choose a voice codec based on the speed/stability of your net-
work as well as the delity of the voice signal you need. To help determine voice load on a
network, in this chapter you will learn how to calculate the size of a frame and bandwidth
consumption based on codec types and sample/payload sizes.
Voice Gateway DSP Functions
Voice gateways do much more than simply route calls between networks. They can also
be used to ofoad processor-intensive tasks from the call-processing agents. Specialized
processors called digital signal processors (DSP) are used to perform multiple voice duties:
PSTN Termination When voice calls must be bridged between an IP network and the
PSTN, trafc is routed to the voice gateway, where a router is used to convert IP voice
packets to PSTN signaling such as a T1 circuit. The conversion requires DSP processing
power to translate between the two networks.
Transcoding Transcoding is the process of translating between two different voice codecs.
There are multiple codecs available for use on voice networks. Codecs are typically chosen
based on hardware compatibility and bandwidth limitations. DSP resources are used in the
translation process, allowing end devices that use different voice codecs to communicate
with each other. A Cisco Unied Communications Manager can perform transcoding
locally, but these can be ofoaded to voice gateways with high-speed DSPs.
Media Termination Point A voice gateway can be congured to be used as a media
termination point (MTP) to relay voice calls that are incoming from either H.323-capable
endpoints or other gateways. An MTP is used to provide endpoints running these signaling
protocols with additional functionality, including:
Call hold
Call transfer
Call park
Conference calling
C05.indd 146 9/21/11 11:18:47 AM
MTPs must also be used in a Cisco environment when there are both SIP and SCCP
phones. SIP DTMF tones are sent inside the payload (in-band), while SCCP phones only
support out-of-band DTMF tones. An MTP can be configured to translate the two tones
between in- and out-of-band.
Conference Calling for Cisco Phones A conference call on a Cisco voice network is
nothing more than the mixing of multiple audio streams (one for each phone in the
conference call) into a single stream that is sent to each phone in the call. In order for this
mixing of audio streams to occur, they must terminate at one point and be processed in near
real time. Similar to transcoding and MTP, a Cisco Unied Communications Manager can
handle some conference-calling duties locally, but doing so is very processor intensive, and
for large implementations its recommended that conference calling be ofoaded to the voice
gateways where DSPs can be used to ofoad call-mixing duties, as shown in Figure 5.1.
In addition, networks using a distributed services deployment model can be configured so
the remote sites voice gateways DSPs are used for local conference calling. This prevents
conference calls from having to needlessly traverse the WAN while consuming bandwidth.
Configuring DSP settings, including DSP farms, will be covered in
Chapter 6, Configuring Voice Gateway Ports and DSPs.
Understanding Voice and VoIP
Quality Considerations
Running voice over an IP network adds some complexity to the task of maintaining the
overall clarity of a call. Because voice is a real-time transmission, network administrators
IP Phone
Voice gateway ofoading
conference calling
IP WAN
V V
M
FI GURE 5.1: Conference call offloading
Understanding Voice and VoIP Quality Considerations 147
C05.indd 147 9/21/11 11:18:48 AM
148 Chapter 5
Chapter
6
c06.indd 179 9/21/11 11:19:30 AM
This is the chapter where we begin to pull in everything
youve learned up to this point about voice networks and voice
gateways and really understand how to congure our Cisco
voice gateways for operation on IP and PSTN networks. In this chapter, well go through
the full conguration process to set up analog and digital interfaces in various scenarios.
In addition, we will go through the process of conguring a digital signal processor (DSP)
farm that ofoads services from a CUCM. At the end of this chapter, we will examine
several show, test, and debug commands used to verify congurations and troubleshoot
voice gateways.
Analog Port Configurations
In this section youll see how to congure FXS, FXO, and E&M ports and dial peers using
various example scenarios, including situations such as PLAR, DID, and CAMA.
Configuring an FXS and an FXO PLAR OPX Port
Our rst example will show how to congure our voice gateway to connect a single FXS port
for an analog telephone with a single FXO port that connects to the PSTN. Because we have
a single phone with a single FXO port, we will use off-premises extension (OPX) Private Line
Automatic Ringdown (PLAR) so that the telephone connected to the FXS interface must be
answered before the FXO interface answers the call, as shown in Figure 6.1.
FI GURE 6.1 An Example of FXS and FXO PLAR OPX
Remote ofce
Ext: 2222
PLAR OPX
2222
555-321-1234
0/1/0
FXO
0/0/0
FXS
V
PSTN
FXS interfaces commonly connect analog telephones or fax machines to voice gateways.
To congure an FXS port, you need to enter into config-voiceport mode by choosing the
slot/port number you wish to congure. For example, if we wanted to congure FXS port
0/0/0 on our router, we would issue the following commands:
c06.indd 180 9/21/11 11:19:31 AM
Analog Port Configurations 181
Router#configure terminal
Router(config)#voice-port 0/0/0
Router(config-voiceport)#
Once we are in config-voiceport mode, the FXS ports can be congured for various
signaling. By default, FXS ports are congured to operate identically to a POTS line in the
United States. Some of the default conguration settings will need to be modied to have the
ports operating properly based on locale. For example, lets say you have a voice gateway
that needs to connect FXS port 0/0/0 for a single analog phone. The phone and voice
gateway are located in Thailand. You should consider modifying the following options:
signal You can change the signaling from the default loopstart to groundstart. Loop-
start signaling has no current owing through it unless it is in use. Therefore it is cheaper
to use and commonly found in residential homes. Ground-start signaling uses an alternate
method to help eliminate glare, as you learned, but also uses more electrical current, which
makes it more expensive to run. Therefore, ground start is more commonly found in
businesses and costs extra. In our example conguration, we will choose to congure loop-
start signaling because it is more common.
cptone This command changes the call progress tones based on the locale of the phone.
You can see the different two-letter ISO-3166 country codes by issuing the cptone ?
command, as shown here:
Router(config-voiceport)#cptone ?
locale 2 letter ISO-3166 country code
AR Argentina IN India PE Peru
AU Australia ID Indonesia PH Philippines
AT Austria IE Ireland PL Poland
BE Belgium IL Israel PT Portugal
BR Brazil IT Italy RU Russian Federation
CA Canada JP Japan SA Saudi Arabia
CN China JO Jordan SG Singapore
CO Colombia KE Kenya SK Slovakia
C1 Custom1 KR Korea Republic SI Slovenia
C2 Custom2 KW Kuwait ZA South Africa
CY Cyprus LB Lebanon ES Spain
CZ Czech Republic LU Luxembourg SE Sweden
DK Denmark MY Malaysia CH Switzerland
EG Egypt MX Mexico TW Taiwan
FI Finland NP Nepal TH Thailand
FR France NL Netherlands TR Turkey
DE Germany NZ New Zealand AE United Arab Emirates
GH Ghana NG Nigeria GB United Kingdom
c06.indd 181 9/21/11 11:19:31 AM
182 Chapter 6
Describe H.323.
Describe SIP.
Describe MGCP.
Chapter
8
c08.indd 281 9/21/11 11:24:35 AM
The new CVOICE 8.0 exam requires that test candidates
understand how to congure a basic voice network using a
Cisco Unied Communications Manager Express router.
In addition, the candidate must understand the infrastructure required to support IP
endpoints. This chapter covers the current options for powering IP phones on a network,
and it shows how to congure VLAN trunks and VLAN voice access ports and
network infrastructure services that support voice, including DHCP, NTP, and TFTP.
The remainder of the chapter covers using the CUCM Express IOS command-line software
to congure and verify voice settings and operational status for both the SCCP and
SIP protocols.
Voice Network Infrastructure
Considerations
There are several network infrastructure factors that you must consider when implementing
a CUCM Express or any voice system over an IP network. In this section we will cover IP
phone power options, voice VLAN congurations, and network services such as DHCP
and NTP that support the use of voice on a network.
Power Options for IP Phones
Cisco IP phones, being much more than simple analog telephones of old, require a power
source to operate. Currently there are three ways of providing power to Cisco IP phones:
Power brick
Powered patch panel/power injector
Power over Ethernet (PoE) switch
Lets briey review each of these IP phone power methods.
Power Brick
The power brick is the simplest to understand. It connects to a power port on the back of
the phone and plugs into a standard 110v AC wall outlet. You then connect a Category 5 or
higher Ethernet cable into a switch to provide network connectivity.
c08.indd 282 9/21/11 11:24:37 AM
Voice Network Infrastructure Considerations 283
The power brick option may be useful in situations where you will use only a handful
of phones. Otherwise, you may want to investigate a PoE option, because it can be more
cost effective and, quite simply, its nicer to combine power and Ethernet in one cable to
eliminate the need for a second connection to the phone.
Powered Patch Panel/Power Injector
A second power option is to have a device that sits between your IP phone and switch
(which is not PoE capable). This is known as a midspan method, because the power
sits in the middle of the connection. A powered patch panel can terminate nonpowered
Ethernet on one end and a powered Ethernet termination point on the other. These patch
panels allow the power to be connected back at the wiring closet, so no power brick is
required and the phone receives both power and Ethernet over a single Category 5 or 6
Ethernet cable.
You can also purchase a Cisco power injector. These devices provide the same midspan
sit-in-the-middle power function as the powered patch panel but only for a single phone
per injector.
Power over Ethernet Switch
The most streamlined and efcient method of providing power to phones (and other
PoE-capable devices) is the Power over Ethernet (PoE) switch. The switch is responsible
for detecting and outputting the required power on each switchport. By adding PoE
functionality to the switch, you have fewer devices that need UPS protection in the event
of a power outage.
There are a couple of gotchas that you need to be aware of when powering Cisco
phones with any PoE option. The rst is to be sure of the type of inline power and quantity
that the phone supports. The second thing to watch out for is ensuring that your switch can
properly handle the power load. Lets look rst at the two inline power methods for Cisco
switches and then at switch power capacities.
Inline Power Method 1: Cisco Inline Power
In its typical fashion, Cisco began offering a proprietary inline-power option to customers
before an open standard was available. In early 2000, Cisco began selling Catalyst switches
with the proprietary inline power (ILP) functionality. ILP uses RJ-45 pins 1, 2, 3, and 6
to provide power to the phones. Using the same wiring that Ethernet uses to transmit and
receive is called phantom power.
Ciscos proprietary inline power provides a xed 6.3W of power to any device that
requires it. ILP detects a capable device by sending a very-low-voltage AC signal across
the transmit pairs and expects to receive the same signal back on the receive pairs.
This is because the ILP-capable phones have a low-pass lter that bridges the specic
voltage signal from TX to RX. Once the switch receives the voltage back on the receive
pair, it knows that the device requires power and sends the 6.3W on that specic
switchport.
c08.indd 283 9/21/11 11:24:38 AM
284 Chapter 8
Chapter
9
c09.indd 353 9/21/11 11:25:35 AM
Were at the point in this study guide where we begin to
expand the capabilities of the IOS voice gateway and CUCM
Express to see what value-added features can be implemented.
In Chapter 9, we will investigate several scenarios that require you to go beyond basic
IP and voice conguration to further enhance the voice experience. We will begin by
conguring DTMF relay support to improve the reliability of DTMF tones on an
IP network. We will then cover the still-important topic of fax machines and modems.
These devices still need to be supported, and youll see how to implement that on
a voice gateway. Finally, youll learn how to implement failover, toll bypass, and
call-restriction techniques.
Configuring DTMF Relay Support
By default, H.323, SIP, and MGCP transport DTMF tones in band. This means the tones
are sent in standard RTP voice packets just as if they were part of a regular voice stream.
This method may work ne for you, but if you are using highly compressed codecs, the
tones may not be reconstructed accurately enough and youll run into connection problems.
For example, when using interactive voice response (IVR) services, it is critical that when
the calling party presses a number to direct them through the IVR menu system, the
number is correctly interpreted by the system so the call can be properly routed.
To make sure that DTMF tones are correctly interpreted, you can congure
DTMF tones to be sent out of band using specially crafted RTP packets, while using a
codec with lower compression to ensure that the digit tones are better replicated at the
opposite end. This section will show how to congure DTMF relay support for H.323,
SIP, and MGCP.
Configuring H.323 DTMF Relay
H.323 DTMF relay is congured while in config-dial-peer configuration mode. To
enable sending of DTMF tones out of band, you simply use the dtmf-relay command
followed by the DTMF method you wish to use. For H.323 the possible relay options
are these:
cisco-rtp This method uses a Cisco proprietary method of transporting DTMF tones in
special RTP packets.
c09.indd 354 9/21/11 11:25:36 AM
Configuring DTMF Relay Support 355
h245-alphanumeric This method uses the H.245 alphanumeric user input method for
specifying only dial-pad tones, namely, 09, *, #, and the AD buttons that are represented
as ASCII characters.
h245-signal This uses the H.245 tone signal method that sends the same dial-pad
tones in ASCII format as the h245-alphanumeric method does. The h245-signal
method also sends along the length of time that the button was pressed, which is
sometimes necessary.
rtp-nte This method uses the named telephone event dened in RFC 2883, which
species a standard method for transporting DTMF tones in RTP packets. One optional
keyword that is compatible with H.323 is digit-drop, which will explicitly drop the in-
band tones from being sent. Without this command, the DTMF tones will be sent both in
and out of band.
The remote end gateway that you are communicating with must also be congured
to use one of these out-of-band signaling methods. When you congure DTMF relay on
a dial peer, you can specify one or more DTMF relay methods. The order of priority is
determined by the router, and the order in which you congure them has no effect. Cisco
rates the order of priority as follows:
1. cisco-rtp
2. rtp-nte
3. h245-signal
4. h245-alphanumeric
As an example of how this works, we will congure VoIP dial peer 100 to use both
H245 alphanumeric and H245 signal methods:
Router#configure terminal
Router(config)#dial-peer voice 100 voip
Router(config-dial-peer)#dtmf-relay h245-alphanumeric h245-signal
Router(config-dial-peer)#end
Router#
So now our voice gateway is congured to use either H.323 alphanumeric or H.245
signal methods. But even though H.245 alphanumeric was entered in the command rst,
the gateway will still prefer to use H.245 signal.
Configuring SIP DTMF Relay
Conguration of DTMF relay using SIP is similar to the H.323 conguration, except that
your dial peer must specically have the session protocol sipv2 command to enable SIP;
by contrast, H.323 is enabled by default. One DTMF relay method is compatible with both
c09.indd 355 9/21/11 11:25:37 AM
356 Chapter 9
Chapter
10
c10.indd 395 9/21/11 11:26:22 AM
In this chapter well look at equipment that is used to help
manage large voice networks. First, well examine the H.323
gatekeeper to see how it can be used to break networks into
zones and how to interact with multiple gatekeepers that control different zones within a
network. You might notice that gatekeepers arent part of the ofcial exam objectives, but
understand that they are a critical part of the 642-437 exam. Once we nish our coverage
of gatekeepers, well move on to look at the Cisco Unied Border Element (CUBE) to see
how it is different from a standard voice gateway and how it can connect two voice net-
works using a pure IP-to-IP solution when the networks are running either SIP or H.323.
What Is an H.323 Gatekeeper?
H.323 can function fairly well on its own just being congured on voice gateways, as you
learned in Chapter 7, Conguring Voice Gateway Signaling Protocols. When you begin
dealing with larger networks, H.323 simply doesnt scale well without the help of an
H.323 gatekeeper to manage your voice network, by breaking it up into multiple zones.
Your H.323 gateways will quickly become cluttered with multiple dial peers that often
cause confusion, and are a pain to maintain when you are dealing with multiple voice
gateways. A better solution is to install one or more gatekeepers into an H.323 network
to perform the following mandatory and optional functions, shown in Table 10.1.
TABLE 10.1 Mandatory and optional H.323 gatekeeper functions
Mandatory Optional
Zone management Call authorization
Address translation Call management
Call admission control (CAC) Bandwidth management
Bandwidth control
Lets break down each of these mandatory and optional H.323 functions to better
understand what the H.323 gatekeeper can provide.
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What Is an H.323 Gatekeeper? 397
H.323 Gatekeeper Mandatory Features
The primary responsibilities of an H.323 gatekeeper are to control call routing, call
permission, and call settings on the network. The H.323 mandatory features control each
of these functions.
Zone Management
Gatekeepers use the concept of logical zones to segment large networks into small and more
manageable chunks. A single zone may contain one or more voice gateways, multipoint
control units (MCUs), or H.323 endpoints. The H.323 gatekeepers responsibility is to
manage all registered devices within the zone and to provide information about how to route
calls between zones. Figure 10.1 shows an example of a gatekeeper managing two different
zones in a network.
Zone1 Zone2
Gatekeeper
V
Voice
Gateway1
V
Voice
Gateway2
V
FI GURE 10.1 A network controlled by a single H.323 gatekeeper
In our example, you see that we have two zones connected to our gatekeeper. The
gatekeeper that directly controls a zone considers them to be local zones, yet our gatekeeper
does not specically belong to a zone itself.
There can also be multiple gatekeepers congured that manage different zones, as shown
in Figure 10.2.
FI GURE 10. 2 A network controlled by multiple H.323 gatekeepers
Zone1 Zone2
Gatekeeper1 Gatekeeper2
V V
Voice
Gateway1
V
Voice
Gateway2
V
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398 Chapter 10
Chapter
11
c11.indd 439 9/21/11 11:27:09 AM
As soon as IP networks were designed and implemented
with sufcient redundancy mechanisms in place to rival
traditional voice systems in stability, it was only a matter
of time before voice made the transition to IP. During this early transition period, early
adopters began noticing that for voice trafc to function as well on a packet network as it
did on traditional circuit-switched networks, the transport method used by IP networks
needed some additional policies and compression techniques in place. Thus began the rise
of Quality of Service (QoS), the collective term for queuing techniques devised to help
eliminate bottlenecked areas on a network.
This chapter covers the who, what, when, where, and why of QoS on IP networks.
Newly added voice trafc began creating bottlenecks, and these bottlenecks led to the
need to create a way to prioritize and queue these packets that are highly sensitive to drops
and latency. Specically, you will learn what it is that causes IP networks to falter when
running real-time streaming voice and video and how QoS and compression techniques can
be used to eliminate each of those problems.
In Chapter 12, Conguring Quality of Service, well move on to the how of QoS on
IP networks as we explore conguring various QoS scenarios.
Problems with Voice/Video
on IP Networks
To understand what QoS does, you need to understand the problems it was introduced
to solve. Before the convergence of time-sensitive transport such as voice and video, IP
networks dealt with applications and data that had the following characteristics:
Large packet payloads
Bursty transport ow
Time-exible transmissions
No one application or data ow with higher priority than another on shared links
The ability to recover in the event of packet drops
As you can see, most data trafc before voice and video were added was inherently
robust. It didnt really matter how long it took for data to get from point A to point B,
as long as it was transported without errors. Thus you see that most data applications
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Mitigating IP Network Voice Issues 441
were built using TCP, which has built-in CRC checks and retransmission of lost or
damaged packets.
Todays modern IP networks that carry voice and video have very different transport
needs outside of the standard data ows just described. Now a network must also provide
mechanisms to carry trafc with these characteristics:
Small packet payloads
Continuous transport ow
Time-sensitive payloads
A way to dene some data ows as higher priority than others on shared links
High sensitivity to packet drops
Because of these new requirements, network administrators must focus on four primary
modications to ensure that voice/video trafc does not suffer on an IP network. Well look
at those factors in the next section.
Mitigating IP Network Voice Issues
Now that converged voice/video and data networks are here to stay, network designers and
administrators must educate themselves about addressing IP network issues so that time-
sensitive data can properly be transported in a reliable and efcient manner. There are four
primary issues to address:
Providing sufcient bandwidth for a converged network
Reducing end-to-end delay
Reducing jitter
Eliminating packet loss
Lets break down each of these issues to see how they can be resolved on a network. You
will then learn how to implement QoS congurations to mitigate the issues in Chapter 12.
Providing Sufficient Bandwidth for
a Newly Converged Network
When planning for a converged voice/video and data network over IP, you must carefully
consider how to allow for the increase in bandwidth usage. There are several considerations
when determining how much bandwidth will increase when adding IP voice to the mix.
These include things such as:
Number of users
Internal versus external calling
Remote site bottlenecks
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442 Chapter 11
Chapter
12
c12.indd 473 9/21/11 11:27:51 AM
Chapter 12 is where the QoS rubber meets the network
road. Chapter 11, Introduction to Quality of Service,
covered the basics of QoS, and now it is time to apply what
you learned to the routers and switches. You will learn how to implement QoS policies
using the AutoQoS methods at Layer 2 and Layer 3, as well as the three-tiered MQC
mechanism, where we mark trafc ow, set policies, and apply them to interfaces.
In addition, we will look at how to congure class-based link efciency techniques, trafc
policing and shaping, trust boundaries, and Layer 2 to Layer 3 mapping modications.
At the end of this chapter, you should have a solid understanding of how to congure key
QoS components as well as how to verify their operation.
Configuring QoS Policies
Using AutoQoS
If you quickly want to get a uniform QoS implementation up and operational on a network,
AutoQoS is the way to go. Essentially, AutoQoS is a built-in script where the router
automatically evaluates a network and then applies QoS settings based on the scripts best
guess at a policy that will work in a particular infrastructure environment. The evaluation
includes verication of interface types and link speeds. The AutoQoS conguration method
is by far the easiest method to implement because little knowledge is required of you in
order to implement it. QoS deployment times are greatly reduced, and the best-practice
congurations are uniform on all network equipment.
AutoQoS can be congured on both routers and switches, although their congurations
and operations vary. Only certain routers are capable of using AutoQoS. The current
generation of ISR and ISR G2 routers supports AutoQoS. From a router perspective,
AutoQoS is commonly congured on WAN interfaces that may be bottlenecks at some
point along a path. AutoQoS may congure the following features on WAN interfaces:
Automatic classication of RTP, cRTP, and voice gateway signaling protocols (SCCP,
H.323, SIP, MGCP)
Automatic building of service policies for priority trafc
LLQ implementation for high-priority trafc
Trafc shaping where appropriate
Link fragmentation where appropriate
cRTP compression where appropriate
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Configuring QoS Policies Using AutoQoS 475
The automatic classification function within AutoQoS uses the Network-
Based Application Recognition (NBAR) feature to identify and classify
different application and data types based on Layer 4 UDP and TCP port
numbers. In order for NBAR to work on a router interface, Cisco Express
Forwarding (CEF) must be enabled first. CEF has been enabled by default,
beginning at IOS 12.2, so if you are running an earlier version, you must
make sure to manually enable it.
On the LAN side of the network, any Cisco Catalyst switch can have AutoQoS for VoIP
congured on its access ports and trunk ports. The following QoS features can be enabled
on switchports using AutoQoS for VoIP:
Set the trust boundary at the Cisco IP phone
Set the trust boundary at the access port or trunk-port level
Automatically enable PQ and WRR queuing when appropriate
Automatically add or modify CoS markings where appropriate
Automatically adjust queue sizes and weights where appropriate
Perform CoS-to-DSCP or IP precedenceto-DSCP mappings
You must choose from two AutoQoS implementation methods when conguring
AutoQoS on a router:
AutoQoS for VoIP
AutoQoS for the Enterprise
The AutoQoS for VoIP is the least complex AutoQoS method, and it primarily
focuses on prioritizing trafc for voice. It can be congured on either routers or switches.
Larger networks with a substantial number of remote site WAN connections may benet
from additional prioritization for trafc types other than voice (such as video and other
streaming applications), and the more complex AutoQoS for the Enterprise is likely to be
a better t. Note that QoS for the Enterprise can be congured only on routers and not
switches. Well start by going through the AutoQoS for VoIP conguration for both routers
and switches followed by conguring AutoQoS for the Enterprise, while pointing out
differences between the two implementation methods along the way.
Configuring AutoQoS for VoIP on a Router
Conguring AutoQoS on routers is truly a magical thing to see. It seems magical because
AutoQoS intelligently recognizes a network setup and appropriately congures multiple
QoS settings. And when I say multiple, I mean it. The AutoQoS for VoIP command is
entered while in config-if mode on any router interface you choose. The command to kick
off the AutoQoS for VoIP process is auto qos voip. There is one optional keyword that
can follow this command, trust. This keyword tells the router to trust the DSCP values
that have been already marked on incoming packets. If the trust keyword is not used, the
router uses NBAR and marks (or re-marks) packets. If you trust your endpoints and their
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476 Chapter 12
Appendix
bapp.indd 529 9/20/11 1:34:45 PM
What Youll Find on the CD
The following sections are arranged by category and
summarize the software and other goodies youll nd on the CD. If you need help with
installing the items provided on the CD, refer to the installation instructions in the Using
the CD section of this appendix.
Sybex Test Engine
The CD contains the Sybex test engine, which includes two bonus practice exams for
Exam 642-437.
Electronic Flashcards
These handy electronic ashcards are just what they sound like. One side contains a
question and the other side shows the answer.
PDF of the Glossary
We have included an electronic version of the Glossary in PDF format. You can view the
electronic version of the book with Adobe Reader.
Adobe Reader
Weve also included a copy of Adobe Reader so you can view PDF les that accompany the
books content. For more information on Adobe Reader or to check for a newer version,
visit Adobes website at www.adobe.com/products/reader/.
bapp.indd 530 9/20/11 1:34:46 PM
Troubleshooting 531
System Requirements
Make sure your computer meets the minimum system requirements shown in the following
list. If your computer doesnt match up to most of these requirements, you may have
problems using the software and les on the companion CD. For the latest and greatest
information, please refer to the ReadMe le located at the root of the CD-ROM.
A PC running Microsoft Windows 98, Windows 2000, Windows NT4 (with SP4 or
later), Windows Me, Windows XP, Windows Vista, or Windows 7
An Internet connection
A CD-ROM drive
Using the CD
To install the items from the CD to your hard drive, follow these steps:
1. Insert the CD into your computers CD-ROM drive. The license agreement appears.
Windows users: The interface wont launch if you have autorun disabled.
In that case, click Start Run (for Windows Vista or Windows 7, Start
All Programs Accessories Run). In the dialog box that appears, type
D:\Start.exe. (Replace D with the proper letter if your CD drive uses a
different letter. If you dont know the letter, see how your CD drive is listed
under My Computer.) Click OK.
2. Read the license agreement, and then click the Accept button if you want to use the CD.
The CD interface appears. The interface allows you to access the content with just one or
two clicks.
Troubleshooting
Wiley has attempted to provide programs that work on most computers with the minimum
system requirements. Alas, your computer may differ, and some programs may not work
properly for some reason.
The two likeliest problems are that you dont have enough memory (RAM) for
the programs you want to use or you have other programs running that are affecting
installation or running of a program. If you get an error message such as Not enough
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532 Appendix