Voice Primer Slides

CCNA Voice Primer
Developed for the Cisco Networking Academy Community by:

Bernard Brunet, Anil Datta, Ben Franske, and Brent Seiling
1
Voice Primer
Table of Contents
1. Understanding Traditional Telephony
2. Introducing Analog Circuits
3. Introducing Digital Circuits
4. Understanding Packetization
5. Introducing VoIP Signaling Protocols
6. Preparing the Network to Support Voice
7. Introducing Cisco Unified Communications Manager Express (CME)
8. Global Telephony Commands
9. Defining Ephone-dn and Ephone
10. Configuring CME to Support Endpoints
11. Dial Peers and Destination Patterns
2
1. Understanding Traditional
Telephony
3
Public Switched Telephony Network
Enterprise
Interoffice
Local Trunk CO
PBX
Loop Trunk
Telephone
CO CO Tie
Telephone
Switch Switch Trunk
PSTN
Enterprise
PBX
Telephone
4
Traditional Business Phone System
PBX Key System
CO CO
Switch Switch
Local Local
Digital Loop Loop
Handsets Analog or
Tie Digital Handsets
Line
Customer
Telephone
5
What Is a PBX?
PBX
Line Card Trunk Card
Terminal Interface Terminal Interface Local

Exchange
Line Switching Network Terminal Interface
Trunk
Control Complex Terminal Interface
Power Supply & Fans Terminal Interface
6
What Is a Key System?
Local
Key System Exchange
Termination
Blocks Trunks
Connector
Block
CO Line Cards
Station Cards
Station Cards
Intercom Cards Main Distribution Frame
Power Supply
7
Comparing Key Systems to PBXs
PBX Key System
Technology Primarily digital Analog or digital
Switch Similar to the CO

Not a switch
Functionality switch
Large company site Small company or branch

Typical
(typically more than 50 office (typically 50 or fewer
Installation
users) users)
Method for Dial 9 or other access

Press a button to access
Accessing number to access
outside line
Outside Trunks outside line
8
Signaling Types
There are three types of signaling used in a

telephony network:
Supervisory signaling communicates the state of a
telephony device.
Address signaling sends information about the digits
dialed.
Informational signaling communicates the current state
of the call.
Signaling can be sent either in-band or out-of-
band.
In-band signaling sends the signaling in the same
communications channel as the voice.
Out-of-band signaling sends the signaling in a separate
communications channel from the voice.
9
Address Signaling
Tone telephone Rotary telephone

DTMF dialing Pulse dialing
10
Signaling System 7
SS7
CO Switch CO Switch
SS7
SS7 is used between telephone companies

SS7 functions:
Informational signaling
Call setup
Call routing
Call billing
Toll-free number resolution
Uses out-of-band signaling
11
PSTN Call Setup
Target
Customer Handset
Telephone PSTN
PBX
Analog Digital or
Circuit CO CO Analog Trunk
Switch Switch
1. Customer phone goes off hook creating a closed circuit.

2. The customers CO switch detects that current is flowing and generates dial tone to the
customer phone.
3. Either DTMF or pulse digits are dialed by the customer.
4. The CO switch collects the digits and performs an SS7 lookup to determine the destination
CO switch.
5. Supervisory signaling indicates to the far-end analog or digital trunk that an inbound call has
arrived.
6. The PBX determines which internal extension the call should go to and causes the target
handset with that extension to ring.
7. Ringback is generated to the customer phone by their local CO switch.
8. The target handset goes off hook and a circuit is built end-to-end.
12
Understanding Traditional Telephony
Summary
The traditional telephony network is composed of the PSTN,

PBXs, key switches, signaling, call setup, and numbering plans.
Placing a call through the PSTN can involve analog circuits,
digital circuits, CO switches, and interoffice trunks.
A PBX is used in larger installations and is similar to a CO switch.
Key systems are used at smaller sites, have fewer features than
a PBX, and the users have shared line appearances on all
phones.
Supervisory signaling communicates state changes in an analog
phone or digital handset; address signaling communicates the
dialed digits using DTMF or pulse; and informational signaling
communicates with the caller or called party.
13
2. Introducing Analog Circuits
14
Components of an Analog Telephone
Receiver
Transmitter
Two-wire/four-wire hybrid
Dialer (DTMF or pulse)
Switch hook
Ringer
15
FXS Interface
FXS
FXS
FXS
Connects directly to analog phones or faxes

Provisions local service
Emulates the CO to the attached devices
Provides power, call progress tones, and dial tone
16
FXO Interface
FXO FXO
PSTN
Connects directly to office equipment

Used to make and receive calls from the PSTN
Can be used to connect through the PSTN to
another site
Answers inbound calls
17
Analog Circuits Summary
Analog phones have a receiver, transmitter, two-

wire/four-wire hybrid, dialer, switch hook, and
ringer.
FXS ports simulate a CO to an analog phone or
fax that is attached to the port.
FXO ports connect a Cisco voice gateway to a
CO switch or to an analog port on a PBX.
Analog circuits include FXS, FXO, and E&M
circuits.
18
3. Introducing Digital Circuits
19
Digitizing Analog Signals
1. Sample the analog signal regularly.

2. Quantize the sample.
3. Encode the value into a binary expression.
4. Compress the samples to reduce bandwidth
(optional).
20
Step 1Sample the Signal
Analog Waveform
Time
Each sample is 1/8000 of a second apart.

21
Step 2Quantize the Signal
Voltage
Segment 2
+ Segment 1
Segment 0
Segment 0
Each sample is 1/8000 of a Time
second apart
- Segment 1
mu-law
Segment 2
22
Digital Circuits Summary
To digitize an analog signal, samples must be

taken regularly, quantized to a binary value,
and may optionally be compressed
T1 and E1 circuits are the most common
digital circuits.
23
4. Understanding Packetization
24
Digital Signal Processors
DSPs
Analog or
Digital IP Packets
PSTN IP
Analog or
Digital IP Packets
PSTN
Analog or
Digital IP Packets
PSTN IP
Speech IP Packets
IP
25
Digital Signal Processors (Cont.)
The DSP chip performs the sampling, quantization,

encoding, and optional compression step of digitization.
It is used in both directions to convert from a traditional
analog or digital voice signal to VoIP; or from VoIP to a
traditional analog or digital voice signal.
The number of simultaneous calls a chip can handle
depends on the type of DSP and the codec being used.
26
Real-Time Transport Protocol
Payload Sequence
Time Stamp Payload
Type Number
Provides end-to-end network functions and delivery

services for delay-sensitive, real-time data, such as
voice and video
Randomly picks even ports from UDP port range
1638432767
Includes the following services:
Payload type identification
Sequence numbering
Time stamping
27
RTP Control Protocol
Can be used to monitor the quality of the data

distribution and provide control information
Provides feedback on current network conditions
Allows hosts that are involved in an RTP session to
exchange information about monitoring and
controlling the session:
Packet count
Packet delay
Octet count
Packet loss
Jitter (variation in delay)
Provides a separate flow from RTP for UDP
transport use
Is paired with its RTP stream and uses the same
port as the RTP stream plus 1 (odd-numbered port)
28
Packetization
Layer 2 IP UDP RTP Header Voice

Header Header Header Payload
Packetization of voice is performed by DSP resources.

The DSP packages voice samples or compressed
voice into
IP packets.
The voice data is collected until the packet payload is
full.
The voice data is carried in the payload of RTP
segments.
RTP is encapsulated in a UDP segment, which is
encapsulated in an IP packet.
The IP packet is encapsulated into the Layer 2 format.
29
G.711 codec example
10010111 Sample 1
10010110 Sample 2
10010101 Sample 3
10010100 Sample 4
10010011 Sample 5
...
10110001
Sample 160
RTP Header 10010111 10010110 10010101 10010100 10010011 ... 10110001
G.711 20 ms of samples (160 bytes)

30
G.729 codec example
10010111 Sample 1
10010110 Sample 2
10010101 Sample 3
10010100 Sample 4
... Sample 160
10110001
DSP Compression
RTP Header 20 Bytes of Voice Payload

G.729 20 ms of voice contained in packet
31
CodecsBandwidth Implications
Codec G.711 iLBC G.729
Bandwidth not 64 13.3 8

including kb/s kb/s kb/s
overhead
G.711, G.729, and iLBC are the most common codecs.
32
Some Additional DSP Functions
Conferencing
Transcoding between two different codecs
Echo cancellation
33
Understanding Packetization Summary
DSP resources are critical to a Cisco Unified Communications

system and translate traditional voice data to IP packets and
back.
Voice is packaged into RTP segments; RTP segments are
encapsulated into UDP segments; UDP segments are
encapsulated into IP packets; and IP packets are
encapsulated into the specific Layer 2 they will traverse.
RTP is used to carry voice and video data across the IP
network, and RTCP is used to provide feedback on the RTP
stream.
The most common codecs used are G.711, G.729, and iLBC.
DSP resources can also provide echo cancellation and call
features such as conferencing and transcoding.
34
5. Introducing VoIP Signaling
Protocols
35
VoIP Signaling Protocols
Signaling generates and monitors the call control
information between two endpoints to:
Establish the connection
Monitor the connection
Release the connection
The signaling protocol must pass supervisory,
informational, and address signaling.
Signaling protocols can be peer-to-peer or client/server-
based.
Peer-to-peer allows the endpoints to contain intelligence to
place calls without assistance.
Client/server puts the endpoint under the control of a centralized
intelligence point.
36
VoIP Signaling Protocols Comparison
Used on Used on Cisco

Architecture
Gateways Unified IP Phones
H.323 Yes No Peer-to-peer
Yes, Cisco Unified

SIP Yes IP Phones and Peer-to-peer
third-party phones
SCCP Yes, Cisco Unified
Yes, limited Client/server
(Skinny) IP Phones only
37
Voice Protocols Example
Voice Voice
Gateway 1 Gateway 2
VoIP ITSP
10.10.10.1 10.10.10.2 192.168.10.1
SCCP VoIP VoIP Destination

(Skinny) H.323 SIP
38
Introducing VoIP Signaling Protocols
Summary
Signaling protocols are used in VoIP networks to

set up new calls, monitor current calls, tear down
calls, pass informational signaling, pass supervisory
signaling, and pass address signaling.
SCCP is a proprietary protocol used between Cisco
Unified IP Phones and Cisco Unified
Communications call control products.
H.323 is a stable, mature, vendor-neutral protocol
that is widely deployed.
SIP is an emerging protocol based on parts of
existing protocols. It is still evolving.
39
6. Preparing the Network to
Support Voice
40
Advantages of Voice VLANs
Phones segmented in
separate logical
networks
Provides network
segmentation and
control
Allows administrators
to create and
enforce QoS
Lets administrators
add and enforce
security policies
41
VLAN Operation
42
Voice VLANs
Separates voice and data traffic

Prevents unnecessary IP address renumbering
Simplifies QoS configurations
Requires two VLANs: one for data traffic and one for
voice traffic
Requires only one Ethernet cable drop for the Cisco
IP phone
and the PC that is plugged into the phone
Requires two IP subnets: one for the data VLAN and
one for the voice VLAN
43
Voice VLANs (Cont.)
An access port can handle two VLANs:

Access VLAN
Voice VLAN
Tagged 802.1Q (Voice VLAN)
Untagged 802.3 (Access VLAN)
44
Configuring Voice VLANs
Console(config)#interface FastEthernet0/1
Console(config-if)#switchport access vlan 12
Console(config-if)#switchport mode access
Console(config-if)#switchport voice vlan 112
Console(config-if)#spanning-tree portfast
The access VLAN is used for the PC that is

plugged into the IP phone.
The voice VLAN is used for voice and signaling that
originates and terminates on the Cisco IP phone.
Spanning-tree PortFast mode causes spanning tree
to enable the port more quickly.
45
Verifying Voice VLAN Configuration
Switch#show interface fa0/17 switchport
Name: Fa0/17
Switchport: Enabled
Administrative mode: static access
Operational Mode: static access
Administrative Trunking Encapsulation: negotiate
Operational Trunking Encapsulation: native
Negotiation of Trunking: Off
Access Mode VLAN: 12 (VLAN0012)
Trunking Native Mode VLAN: 1 (default)
Voice VLAN: 112 (VLAN0112)
Trunking VLANs Enabled: ALL
Pruning VLANs Enabled: 2-1001
Appliance trust: none
46
DHCP Service
Assigns IP addresses and subnet masks for

one or more subnets
Assigns a default gateway
Assigns DNS servers (optional)
Assigns other commonly used servers
(optional)
Needs to be customized to assign a TFTP
server to the voice VLAN that IP phones are on
Configure a separate DHCP scope for the IP
phones as a best practice
47
Phone Bootup
The IP phone powers on.
The phone performs a POST.
The phone boots.
The phone uses Cisco Discovery

Protocol to learn the voice VLAN.
The phone initializes the IP stack.
48
Phone Bootup (Cont.)
The IP phone sends a

broadcast requesting an IP
address.
The DHCP server selects a free IP

address from the pool and sends it,
along with the other parameters,
including option 150.
The IP phone initializes, applying the
IP configuration to the IP stack.
The IP phone requests a

configuration file from the TFTP
server defined in option 150.
The configuration file will contain

the IP address of the call agent to
register to.
49
Configuring DHCP Service (Cont.)
CMERouter(dhcp-config)#
default-router IP-address
Sets the default gateway that is assigned to DCHP clients
dns-server primary-IP [secondary-IP]
Sets the DNS server or servers that are assigned to the DHCP
clients (optional)
option 150 ip IP-address
Defines the TFTP option and what TFTP server to assign to the
clients
50
Configuring DHCP Example
CMERouter(config)#ip dhcp excluded-address 10.112.0.1 10.112.0.10
CMERouter(config)#ip dhcp pool mypool
CMERouter(dhcp-config)#network 10.112.0.0 255.255.255.0
CMERouter(dhcp-config)#option 150 ip 10.112.0.1
CMERouter(dhcp-config)#default-router 10.112.0.1
CMERouter(dhcp-config)#dns-server 10.100.0.1 10.100.0.2
CMERouter(dhcp-config)#exit
Option 150 informs the IP phone of the TFTP

servers IP address.
The TFTP server contains the configuration files and firmware for the
IP phone.
51
Network Time Protocol
Correct clock synchronization is important for

performance, troubleshooting, and CDRs.
Each Cisco device has an internal system clock
that can be set from a number of sources, such
as an internal calendar system and NTP.
NTP allows network devices to synchronize to a
clock master.
The local NTP server can have an attached
clock or can synchronize with a more
authoritative source.
There are free NTP servers available on the
Internet.
52
Network Time Protocol (Cont.)
The IP phone gets its displayed time from the

call control platform to which is registers.
Cisco Unified Communications Manager
Cisco Unified Communications Manager Express
The time of the internal clock of the Cisco
Unified Communications call control platform
should be synchronized with an NTP server.
The time of the Cisco Unified Communications
call control platform is used to stamp all syslog
and trace messages.
53
Configuring the Time
Router(config)#
clock timezone zone hours-offset
Sets the local time zone
Router(config)#
clock summer-time zone recurring [start-date end-date]
Specifies daylight saving time
Router(config)#
ntp server ip-address
Allows the clock on this router to be synchronized with the specified

NTP server
54
Example of Router Set to PST with Daylight
Saving Time Enabled
NTP
Server
10.1.2.3
Cisco Unified Communications IP phone time comes from the

Manager Express router Cisco Unified Communications
synchronizes time with the NTP Manager Express router.
server.
Router(config)#clock timezone pst -8

Router(config)#clock summer-time zone recurring second sunday march
02:00 first sunday november 02:00
Router(config)#ntp server 10.1.2.3
55
Preparing the Network to
Support Voice Summary
Voice VLANs are used to separate voice traffic from

data traffic.
Voice VLANs are configured on the interfaces of the
switch into which the IP phone connects.
NTP allows you to synchronize your Cisco Unified
Communications Manager Express router to a
single clock on the network.
56
Summary (Cont.)
The IP phone requests the firmware, configuration,

and language files when it boots.
The IP phone uses TFTP DHCP option 150 to
download the configuration file which is needed to
register with the call control device. The IP phone
uses its MAC address as part of a created filename
which uniquely identifies the phone. This
configuration file contains the version of firmware to
use and the IP address and port to which the phone
will register.
57
7. Introducing Cisco Unified
Communications Manager
Express (CME)
58
CME Key Features and Benefits
Supports deployments of up to 240 phones on a

single router
Extends capabilities to the small office that were
previously available only to larger enterprises
Is based on Cisco IOS Software
Can be administered by GUI or CLI
59
Supported Platforms
Cisco Unified Communications Manager
Express supports these Cisco platforms:
Cisco 2800 & 2900 Series
Integrated Services Routers
Cisco 3800 & 3900 Series
Integrated Services Routers
Cisco Unified Communications 500 Series
for Small Business
60
Examples of Cisco Unified IP Phones
Cisco Unified Cisco ATA 186

IP Phone and 188
7942G
Cisco Unified
Wireless IP Cisco Unified IP
Phone 7920 Phone 7962G
61
8. Global Telephony Commands
62
Global Telephony Commands
At a minimum, the router needs to know:
the maximum number of phones allowed
the maximum number of phone numbers to be assigned
the IP address the router uses to respond
The phones also need a default template file created.
Enters CME global config mode
telephony-service Optional command that prevents problems with
phones auto registering
no auto-reg-ephone
Mandatory commands which
max-dn 12 define max number of
extensions, and max number of
max-ephone 8 phones
ip source-address 192.168.0.1 port 2000 Mandatory command to assign

address for router to respond to
create cnf-files phone requests
Mandatory command to build XML
template file for phones
63
Configuration for each phone number
Ephone-dn
Represents the directory number (i.e. the phone number or extension).
The number of extensions is limited by the router model and the max-dn
command.
Must have a directory number assigned before anything else. (PT only
supports directory numbers.)
On real equipment, ephone-dns can be single-line, dual-line, or octal-
line. Packet Tracer only support single-line.
ephone-dn 1 Creates directory number 1
number 1000 Assigns phone number 1000 to directory number 1
ephone-dn 2 Creates directory number 2

number 1001 Assigns phone number 1001 to directory number 2
64
Configuration for each phone
Ephone
Represents the physical IP phone
Must have a mac-address assigned before anything else.
Ephone-dn(s) can be tied to a phone using the button command. On
real equipment, more than one ephone-dn can be used on a phone.
ephone 1 Creates phone 1

mac-address 0001.974c.ae56 Assigns the MAC address of the phone
button 1:1 Assigns phone line button 1 to directory number 1
(which has extension 1000)
ephone 2 Creates phone 2

mac-address 0004.9a2d.2c7c Assigns the MAC address of the phone
button 1:2 Assigns phone line button 1 to directory number 2
(which has extension 1001)
65
9. Defining Ephone-dn and
Ephone
66
Ephone and Ephone-dn Concepts
Ephones and ephone-dns are modular Cisco IOS Software

constructs.
An ephone represents the configuration and setting of the
physical phone.
The maximum number of supported ephones is
determined by the license and hardware platform. Cisco
Unified Communications Manager Express supports a
maximum of 240 ephones.
An ephone-dn is a numeric destination that can be
associated with one or more ephones.
An ephone can have more than one ephone-dn associated
with it.
The maximum number of extensions is the same as the
maximum number of ephone-dns.
67
Ephone-dn Features
An ephone-dn has a primary

Primary extension number on
directory number assigned to a single-line ephone-dn that DN1
it and can have an optional can make or receive one call
at a time
secondary number.
A dn-tag is a unique value that Ephone-dn
is assigned when the ephone-
Primary and secondary
dn is created. extensions configured on a
single-line ephone-dn.
An ephone-dn can be single DN1 and DN2
You need a dual-line ephone-
line or dual line. dn to support call waiting,
consultative transfer, and
A single line can terminate conferencing
one call at a time. Ephone-dn
A dual line can terminate
two simultaneous calls.
Packet Tracer only supports
single line dns
68
Configuring an Ephone-dn
CMERouter(config)#
ephone-dn dn-tag
Creates an extension (ephone-dn) for a Cisco IP phone line
CMERouter(config-ephone-dn)#
number dn-number
Associates a directory number with the ephone-dn instance
69
Basic Ephone-dn Configuration
One Virtual
Voice Port
One Line or 1001

Channel
CMERouter(config)#ephone-dn 7
CMERouter(config-ephone-dn)#number 1001
Assigns a primary extension number to an ephone-dn
70
max-dn Command
CMERouter(config-telephony)# max-dn max-dn
This command sets the maximum definable number of ephone-dns

that can be configured in the system.
The maximum number of supported ephone-dns is a function of the
license and the hardware platform.
The default is 0.
To make the most efficient use of memory, do not set this parameter
higher than needed.
71
max-dn Command (Cont.)
DN DN
DN DN
DN DN
CMERouter(config-telephony )#max-dn 10
DN DN
Attempts to create an
11th ephone-dn will fail. DN DN
72
Ephone Features
IP Phone 7960
An ephone is a software
Button 1 DN Button 4 DN
configuration of a physical phone.
It is assigned a unique phone-tag.
The physical device can be an IP
MAC 000F.2470.F92A
phone or an analog phone attached
to an ATA. IP Phone 7912
The MAC address of the IP phone or
Button 1 DN
ATA is used to tie the software
configuration to the hardware.
You can associate one or more MAC 000F.2470.F92B
ephone-dns with an ephone. Cisco ATA 188
Analog 1 DN
The number of line buttons varies
MAC 000F.2470.F92D
based on the model of phone.
Analog 2 DN
MAC 000F.2470.F92E
73
max-ephone Command
CMERouter(config-telephony)#
max-ephones max-ephones
This command sets the maximum definable number of ephones that

can be configured in the system.
The maximum number of supported ephones is a function of the
license and the hardware platform.
The default is 0.
To make the most efficient use of memory, do not set this parameter
higher than needed.
74
max-ephone Command (Cont.)
CMERouter(config-telephony )#max-ephones 4
4
Attempts to create a
fifth ephone-dn will fail.
75
Configuring an Ephone
router(config)# ephone phone-tag
Creates an ephone instance and enters ephone subconfiguration

mode
router(config-ephone)#mac-address mac-address
Associates the defined MAC address of the physical device with the
ephone
76
Configuring an Ephone (Cont.)
router(config-ephone)#button button-number {separator} dn-tag [[button-

number {separator} dn-tag]]
Associates the ephone-dn(s) with a specific button(s) on
the IP phone
77
Some Button Separators
: Normal ring
b Beep but no ring
f Feature ring
s Silent ring
78
Example: Basic Ephone Configuration
MAC 000F.2470.F8F8
ephone 1
ephone-dn 7:
1001
One Virtual Port
Button 1
000F.2470.F8F8
CMERouter(config-ephone-dn)#exit
CMERouter(config)#ephone 1
CMERouter(config-ephone)#mac-address 000F.2470.F8F8
CMERouter(config-ephone)#button 1:7
79
Multiple Ephones
1004
1004
1005
1005
Four physical phones 1006

1006
Four ephones defined
Four ephone-dns defined
1007
1007 Cisco ATA 186 or
188
80
Example: Configuration for Multiple Ephones
CMERouter(config-ephone)#mac-address 000F.2470.F8F1
CMERouter(config-ephone)#mac-address 000F.2470.A302
CMERouter(config-ephone)#mac-address 000F.2470.66F6
CMERouter(config-ephone)#mac-address 000F.2470.7B54
81
Single-Line Ephone-dn
One Virtual
Voice Port
One Channel 1001
The ephone-dn creates one virtual voice port.

Only one call to or from this ephone-dn can occur at
any one time.
82
Defining Ephone-dn
and Ephone Summary
Ephone-dns and ephones are two key components of

the Cisco Unified Communications Manager Express
system.
An ephone-dn is a single instance of an extension
(directory) number.
An ephone is a single instance of the configuration of
the physical instrument.
83
10.Configuring CME to Support
Endpoints
84
Telephony Service Configuration
CMERouter(config)#
telephony-service
Enters telephony-service mode
max-ephone maximum-ephones
Sets the maximum number of ephones that may be defined in the

system (default is 0)
max-dn maximum-directory-numbers
Sets the maximum number of ephone-dns that may be defined in

the system (default is 0)
85
Source IP and Port
ip source-address ip-address [port port]
Identifies the address and port through which IP phones communicate

with Cisco Unified Communications Manager Express
10.90.0.1
telephony-service
ip source-address 10.90.0.1 port 2000
86
Automatic Registration
auto-reg-ephone
Enables automatic registration of new ephones that are not in the

configuration and is enabled by default
10.90.0.1
telephony-service
no auto-reg-ephone
87
Setup of Cisco Unified Communications Manager
Express from the CLI
tftp-server flash:CP7921G-1.0.3.LOADS
tftp-server flash:APPS-1.0.3.SBN
tftp-server flash:GUI-1.0.3.SBN
tftp-server flash:SYS-1.0.3.SBN
tftp-server flash:TNUX-1.0.3.SBN
tftp-server flash:TNUXR-1.0.3.SBN
tftp-server flash:WLAN-1.0.3.SBN telephony-service
telephony-service
load 7921 CP7921G-1.0.3
create cnf-files
max-ephones 10
max-dn 10
dialplan-pattern 1 2095559... extension-length 4 extension-pattern 1...
See the Defining ephone-dn 1 dual-line
Ephone-dn and number 401
Ephone lesson ephone 1
for configuration mac-address 000F.2745.2AD8
information. button 1:1
88
IP Phone Firmware and XML Configuration
Files
Certain files are necessary

for proper operation of a
Cisco Unified IP phone:
Firmware
XMLDefault.cnf.xml
SEPAAAABBBBCCCC.cnf.x TFTP Server
ml
(where AAAABBBBCCCC is
the
MAC address of the device )
89
Create XML Files
create cnf-files
Builds the specific XML files that are necessary for the IP phones
SEP000F2473AB14.cnf.xml
000F.2473.AB14
10.90.0.1
telephony-service
create cnf-files
90
Registration Flow Chart
Does Is
Phones XML Yes firmware
Yes The phone
file exist? registers
current?
No No
Use default Update Phone
XML file firmware Restarts
Auto Register
Yes Auto assign Yes using a DN
registration
configured? in the pool
enabled?
No No
Phone Register
Restarts without a DN
91
Device Configuration XML File
<device>
<devicePool>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
SEPAAAABBBBCCCC.cnf.xml* <ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName>10.15.0.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<versionStamp>{Jan 01 2002 00:00:00}</versionStamp>
<loadInformation>P0030702T023</loadInformation>
- <userLocale>
<name>English_United_States</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>United_States</networkLocale>
<idleTimeout>0</idleTimeout>
*AAAABBBBCCCC = the <authenticationURL />
MAC address <directoryURL>http://10.15.0.1/localdirectory</directoryURL>
<idleURL />
<informationURL />
<messagesURL />
<proxyServerURL />
<servicesURL />
</device> 92
Default XML File
<Default>
<callManagerGroup>
<members>
<member priority="0">
XMLDefault.cnf.xml <callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName>10.15.0.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation6 model="IP Phone 7910">P00403020214</loadInformation6>
<loadInformation124 model="Addon 7914"></loadInformation124>
<loadInformation9 model="IP Phone 7935"></loadInformation9>
</Default>
93
Automated Deployment of Endpoints
In an automated setup you do not have to configure

ephones.
The automated setup automates the deployment of IP
phones.
Use the auto assign command in telephony service
configuration mode to perform the automatic
assignment.
All of the ephone-dns you want to deploy must be the
same type
(single-line or dual-line).
94
auto assign Command
auto assign start-dn to stop-dn [type phone-type]
Ephone-dns are automatically assigned to new

ephones that are configured.
Phones can take up to five minutes to register.
Wait for all phones to register before saving the
configuration.
95
Example: auto assign Command
New Phone Plugs In
When a new IP phone registers with a Cisco Unified

Communications Manager Express system, a new telephony-service
ephone is created with the MAC address of the IP auto assign 1 to 10 type 7920
phone.
auto assign 11 to 20 type 7940
An existing ephone-dn is assigned to the new ephone
from the range defined for the type of phone. auto assign 21 to 40 type 7960
The lowest unassigned ephone-dn in the matching auto assign 41 to 50
statement range is used.
...
If all ephone-dns in a range have been assigned,
some phones may not receive an ephone-dn or may ephone-dn 1
receive an ephone-dn from the auto assign command number 1000
without a type.
...
If a new IP phone does not match any auto assign
command with a type, the auto assign command
without a type is used.
96
Verify Cisco Unified Communications Manager
Express Phone Configuration
CMERouter1#show running-config
telephony-service
max-ephones 10
max-dn 10
auto assign 1 to 10
create cnf-files
!
ephone-dn 1
number 9000
!
ephone 1
mac-address 000F.2470.F8F8
button 1:1
97
11.Dial Peers and Destination
Patterns
98
Gateways
Translate between
different networks
Require DSP resources
to perform the translation
Can be analog gateways:
Analog station gateways
Analog trunk gateways
Can be digital gateways
99
Gateway FunctionExample on Cisco Unified
Communications Manager Express
Digital
Gateway
T1
PSTN
FXO
Analog
Gateway
100
Voice Ports
Analog ports
FXS
FXO
Digital ports
CAS T1/E1
PRI T1/E1
BRI
101
Call Legs
Phone 1234 dials

a PSTN destination
Router1 Router2
VoIP PSTN
FXS
1/0/1 10.10.10.1 10.10.10.2
Call Leg 1: Call Leg 2: Call Leg 3: Call Leg 4:

In on Out on In on Out on
Router1 Router1 Router2 Router2
102
Dial Peers
Dial peers are an addressable call endpoint.
They establish logical connections, or call legs, to complete an
end-to-end call.
You can use dial peers inbound, outbound, or both.
Dial peers define the properties of the call leg:
Codec
QoS markings
VAD
Fax rate
Cisco voice-enabled routers typically use two types of dial peers:
POTS dial peersconnect to a traditional telephony network
such as FXO, FXS, E&M, BRI, PRI T1/E1, and CAS T1/E1
VoIP dial peersconnect over an IP network using an IP
address
103
Dial Peers (Cont.)
Analog Voice-Enabled Voice-Enabled

Destination Router Router
IP Network
POTS VoIP
You create dial peers using the CLI.
104
POTS Dial Peers
1234
FXS 1/0/1
Dial peer 20 will be used to match

outbound when the router receives a
call setup message for 1234.
CMERouter(config)#dial-peer voice 20 pots

CMERouter(config-dialpeer)#destination-pattern 1234
CMERouter(config-dialpeer)#port 1/0/1
105
Destination Pattern Options
Common destination pattern wildcards:

Plus (+)
Preceding digit occurs one or more times
Asterisk (*) and pound sign (#)
Not valid wildcards; are DTMF tones
Comma (,)
Inserts a one-second pause
Period (.)
Specifies any one wildcard digit
Square brackets
Indicates a range of digits within the brackets
T
Indicates a variable-length pattern
106
VoIP Dial Peers (not supported in PT)
Dial peer 20 Dial peer 30

matches matches outbound
inbound
Phone 1234 CMERouter CMERouter 2010
dials 2010 1 2
VoIP
FXS
1/0/1
Lo0 - Lo0 -
10.10.10.1 10.10.10.2
CMERouter1(config)#dial-peer voice 20 pots

CMERouter1(config-dialpeer)#destination-pattern 1234
CMERouter1(config-dialpeer)#port 1/0/1
CMERouter1(config)#dial-peer voice 30 voip
CMERouter1(config-dialpeer)#destination-pattern 2...
CMERouter1(config-dialpeer)#session target ipv4:10.10.10.2
107
VoIP Dial Peers (Cont.)
Dial peer 40 Dial peer 50

matches inbound matches outbound
Phone 1234
dials 2010 CMERouter1 CMERouter2 2010
VoIP
FXS 1/0/1 FXS 1/1/1
10.10.10.1 10.10.10.2
CMERouter2(config)#dial-peer voice 50 pots

CMERouter2(config-dialpeer)#destination-pattern 2010
CMERouter2(config-dialpeer)#port 1/1/1
CMERouter2(config)#dial-peer voice 40 voip
CMERouter2(config-dialpeer)#destination-pattern 1...
CMERouter2(config-dialpeer)#session target ipv4:10.10.10.1
108
Matching Outbound Dial Peers
Destination pattern is matched based on longest number match.
dial-peer voice 1 voip

destination-pattern .T
session target ipv4:10.1.1.1

destination-pattern 555[2-3]...

destination-pattern 5551...

destination-pattern 5551234
Example 1: Dialed number 555-1234 will match dial peer 4.

109
Internet Telephony Service Providers
ITSPs provide cost savings.

The cost per line is less than traditional offerings.
The long distance charges are lower.
You can purchase lines in increments of one instead
of larger blocks found in E1s, T1s, and PRI.
When not in use for voice, you can use the unused
bandwidth from the connection for other applications.
SIP is the most common protocol used by ITSPs.
To implement, create a VoIP dial peer with the correct
settings for the ITSP to which you are connecting.
110
Internet Telephony Service Providers (Cont.)
Enterprise Network
ITSP
Network
VoIP Dial
Peer
VoIP PSTN
111
Dial Peers and Destination Patterns Summary
Gateways translate between two different networks. They

can be analog or digital.
Voice ports are used to terminate a traditional telephony
interface on the voice gateway.
Call legs represent segments in the call path that connect
two devices.
Dial peers represent programming on the voice gateway
that defines what to do when a call setup message is
received.
An ITSP trunk is an IP connection to the carrier for PSTN
calls.
You can use show commands to verify dial peer and dial
plan configurations.
112
References
Implementing Cisco IOS Unified

Communications, 2008 Cisco Systems, Inc.
(source of most of the graphics)
113
Acknowledgements
Team Members Academies
Bernard Brunet Cgep de lOutaouais
Montgomery County Community

Anil Datta
College
Ben Franske Inver Hills Community College
Brent Sieling Madison Area Technical College
114
Thank you
115

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CCNA Voice Primer

Developed for the Cisco Networking Academy Community by:

PBX Key System

Line Card Trunk Card

Terminal Interface Terminal Interface Local

PBX Key System

Technology Primarily digital Analog or digital

Switch Similar to the CO

Large company site Small company or branch

Method for Dial 9 or other access

There are three types of signaling used in a

Tone telephone Rotary telephone

SS7 is used between telephone companies

1. Customer phone goes off hook creating a closed circuit.

The traditional telephony network is composed of the PSTN,

Connects directly to analog phones or faxes

Connects directly to office equipment

Analog phones have a receiver, transmitter, two-

1. Sample the analog signal regularly.

Each sample is 1/8000 of a second apart.

To digitize an analog signal, samples must be

The DSP chip performs the sampling, quantization,

Provides end-to-end network functions and delivery

Can be used to monitor the quality of the data

Layer 2 IP UDP RTP Header Voice

Packetization of voice is performed by DSP resources.

RTP Header 10010111 10010110 10010101 10010100 10010011 ... 10110001

G.711 20 ms of samples (160 bytes)

RTP Header 20 Bytes of Voice Payload

Codec G.711 iLBC G.729

Bandwidth not 64 13.3 8

G.711, G.729, and iLBC are the most common codecs.

DSP resources are critical to a Cisco Unified Communications

Used on Used on Cisco

H.323 Yes No Peer-to-peer

Yes, Cisco Unified

10.10.10.1 10.10.10.2 192.168.10.1

SCCP VoIP VoIP Destination

Signaling protocols are used in VoIP networks to

Separates voice and data traffic

An access port can handle two VLANs:

Tagged 802.1Q (Voice VLAN)

Untagged 802.3 (Access VLAN)

The access VLAN is used for the PC that is

Switch#show interface fa0/17 switchport

Assigns IP addresses and subnet masks for

The IP phone powers on.

The phone performs a POST.

The phone boots.

The phone uses Cisco Discovery

The phone initializes the IP stack.

The IP phone sends a

The DHCP server selects a free IP

The IP phone requests a

The configuration file will contain

Sets the default gateway that is assigned to DCHP clients

Option 150 informs the IP phone of the TFTP

Correct clock synchronization is important for

The IP phone gets its displayed time from the

Sets the local time zone

Specifies daylight saving time

Allows the clock on this router to be synchronized with the specified

Cisco Unified Communications IP phone time comes from the