Ccna Voice

.
CCNA Voice Introduction

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Houdasite
sharif_soper
lumark_s
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Cisco Unified Communications Manager

Signaling and Media Paths
Express
Signaling Protocol
Signaling Protocol
(SCCP / SIP)
(SCCP / SIP)
IP Phone
A
Media Exchange (RTP)

performs call setup using a Signaling Protocol
(SCCP/SIP).
Media exchange occurs directly between
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endpoints using RTP.
et
IP Phone
B
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Gateway Hardware Platforms (Cont.)

Special voice gateways:
Cisco VG224 and VG248 Gateways
Cisco AS5850 Gateway

Cisco AS5300 and AS5400 Series Gateways
Cisco 827-4V Router

EOS: 05/2005
EOL: 05/2010
Cisco ATA 186
Cisco 7200 Series Routers
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Gateway Hardware
Platforms
Modern enterprise models:
Cisco Catalyst 6500 Series
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GATEKEEPE
R
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Cisco Catalyst Family of

Switches
Cisco Catalyst 6500
Cisco Catalyst 4500
Cisco Catalyst 3750
Cisco EtherSwitch Network Module

Cisco Catalyst 3560
Three Ways to Power Cisco IP

Phones
Power over Ethernet (PoE):
Needs PoE line cards or PoE ports
for Cisco Catalyst switches
Delivers 48V DC over data pairs
(pins 1, 2, 3, and 6) or spare pairs
(pins 4, 5, 7, 8)
Midspan power injection:
Needs external power source
equipment
Delivers 48V DC over spare pairs
Wall power:
Needs DC converter to connect a
Cisco IP phone to a wall outlet
Power
Power Injector
No Power
AC
Source
Power
110 V AC
Wall Power
to 48 V DC
Converter
Two Types of PoE Delivery

Cisco original implementation:
Provides -48V DC at up to 6.3 to 7.7 W per port over data pins
1, 2, 3, and 6.
Supports most Cisco devices (Cisco IP phones and wireless
access points).
Uses a Cisco proprietary method of determining if an attached
device requires power. Power is delivered only to devices that
require power.
IEEE 802.3af Power over Ethernet:

Specifies 48V DC at up to 15.4W per port over data pins 1, 2,
3, and 6 or spare pins 4, 5, 7, and 8.
Enables a new range of Ethernet-powered devices because of
increased power.
Standardizes the method of determining if an attached device
requires power. Power is delivered only to devices that require
power.
Has several optional elements,
including power classification.
Cisco Prestandard Device

Detection
Cisco Prestandard
Implementation
Powered Device Port
Pin3
FLP
Switch
It is an inline device.
Pin6
Rx
FLP
Pin1
Pin2
Tx
IEEE 802.3af Device

Detection
IEEE 802.3af Powered Device
IEEE 802.3af PSE
2.8V to 10V
Switch
It is an IEEE
powered device.
Pin3
Detect Voltage Pin6
Rx
Pin1
Pin2
Tx
25K Ohm
Resistor
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Redundancy Design
Primary
Secondary or
Backup
7500 IP phones
Cisco MCS 7845
Publisher and
TFTP Server
(Not Req.
<1000)
Primary
1 to 7500
15,000 IP phones
Cisco MCS 7845
Publisher and
TFTP Server
Backups
1 to
7500
30,000 IP phones
Cisco MCS 7845
Publisher and
TFTP Server
Backups
7501 to
15,000
1 to
7500
7501 to
15,000
Backup
Backups
15001 to
22,500
22,501 to
30,000
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UCCX SERVER
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Broadest Choices Among IP

Cisco IP
Endpoints
Phone 7970G &
Cisco IP Phone
7960G
7971G
Cisco IP Phone
7940G
Cisco
Wireless
IP Phone
7920
Cisco IP Phone 7905G
and Cisco IP Phone
7912G
Cisco VT Advantage
Cisco IP
Communicator
Cisco IP
Conference
Station 7936
Cisco IP Phone 7902G
Cisco VG248
Analog
Phone Gateway
Cisco ATA 186/188
Cisco Unified Communications

Manager Endpoint Overview
Cisco Unified
Communications
Manager 6.0
Cisco Unified
IP Phones
SCCP Video
Phones
SCCP
SIP
H.323
Cisco SCCP-Only
Phones
Analog
Station Gateways
Third-Party
SIP Endpoints
Third-Party
H.323 Endpoints
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Entry-Level Cisco IP Phones

Cisco Unified
IP Phone 7906
Cisco Unified
IP Phone 7911
Basic-featured Cisco IP phones for low-to-medium

telephone use
Single line or directory number
Message waiting indicator
Midrange Cisco IP Phones

Cisco Unified IP Phone 794[012]
7940
7941
Cisco Unified IP Phone

796[012]
7960
7961
Full-featured Cisco IP phones for medium-to-high telephone use

Multiline
Message waiting indicator
Large pixel-based displays
Integrated switches
Built-in headsets and high-quality speakerphones
Upper-End Cisco IP Phones

Cisco Unified
IP Phone 7945
Cisco Unified
IP Phone 7965
Cisco Unified
IP Phone 797[015]
7970
7975
Addresses needs of executives

Large, color, pixel-based displays
Touch-sensitive display on 797[015] models
Two to eight telephone lines, or combinations of lines and
direct access to telephony features
Four or five interactive softkeys
Built-in headsets and high-quality, hands-free speakerphones
Other Cisco IP Phones

Cisco Unified
IP Phone 7985
Cisco Unified IP
Conference Station 7936
Cisco Unified
Wireless IP Phone
792[01]
7921
Cisco Unified IP
Phone 7931
H.323 Endpoints
Cisco Unified IP Phone 7905 can be loaded with an H.323
firmware.
From Cisco Unified Communications Manager perspective,
they look like any other (third-party) H.323 endpoint.
Other commonly used H.323 phones are Microsoft Windows
NetMeeting or H.323 video devices from vendors like
Tandberg or Sony.
Cisco 7905 IP Phone
Third-Party H.323 Endpoints
Third-Party SIP Phones

Cisco Unified IP Phones 7940 and 7960 can be loaded with
a standard SIP software, which is different from using SIP
with Cisco Unified Communications Manager extensions on
these phones.
From Cisco Unified Communications Manager perspective,
these phones look like any other (third-party) SIP endpoints.
Many third-party SIP phones are available on the market.
Cisco 7960 IP Phone
Third-Party SIP Endpoints
Cisco IP Phone Startup Process

DHCP
Unified CM
Cisco TFTP
6
4
2
1.
2.
3.
4.
5.
6.
Cisco IP phone obtains power from the switch

Cisco IP phone loads locally stored image
Switch provides VLAN information to Cisco IP phone using Cisco
Discovery Protocol
Phone sends DHCP request; receives IP information and TFTP server
address
Cisco IP phone gets configuration from TFTP server
Cisco IP phone registers with Cisco Unified Communications Manager
server
DHCP Discover
DHCP Offer
DHCP Request
DHCP ACK
Contains: option 150 or option 66
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Files
Files critical to the IP phone
Firmware
XML
SEPAAAABBBBCCCC.cnf.xml
XmlDefault.cnf.xml
TFTP Server
SCCP-dictionary.xml
Phonemodel-dictionary.xml
Phonemodel-tones.xml
SEP
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Loop Start
When the FXO is idle, the two-wire local
loop is open. The tip and ring leads are not
connected to each other. The ring lead
connects to a - 48 VDC battery at the FXS
side. The tip lead connects to ground at the
FXS side. No current is flowing in the idle
state.
To seize the circuit, the FXO simply
connects the ring wire to the tip wire by
going off-hook. The FXS side detects
current flowing from battery on the ring
lead to ground on the tip lead. The FXS
returns dial tone to the FXO. The FXO then
sends the dialed digits.
If the FXS wants to seize the

circuit to deliver an inbound call,
it applies a 90 VAC ring voltage
on top of the DC voltage on the
ring lead. After seeing this ring
voltage, the FXO may then
connect the ring lead to the tip
lead to accept the inbound call.
The FXS detects that the FXO has
closed the circuit because
current flows from battery to
ground. In response, the FXS
removes the ring voltage and
completes the call.
Glare
In the US for example, ring voltage is sent
for two seconds and not sent for the next
four seconds. When the FXS signals an
inbound call, the cadence does not always
start with the two-second 'on' phase.
Consequently, the FXS may think it has
seized the circuit, but the FXO may not
detect the signal for up to four seconds.
During this four-second window, the FXO
may attempt to seize the circuit for an
outbound call. If both sides seize the circuit
at the same time, the resulting condition is
called 'glare.'
When the FXO is a telephone,

glare is not a big problem. Often a
residential phone user will take
the phone off-hook to place an
outbound call only to find
someone already on the phone.
Glare is a problem on a residential
phone only if the end user is
trying to avoid the inbound caller.
When the FXO is a PBX or router
trunk link, it may be shared by
many end users at different times.
Glare in this type situation may
make a business seem
unprofessional. Inbound
callers
Ground Start
When idle, the FXS attaches the ring lead to
battery, but does not attach the tip lead to
ground. The FXO connects the tip lead to a tip
ground detector circuit. During the idle state, the
ring lead in the FXO is not attached to anything.
If the FXS side wants to seize a ground-start

trunk for inbound call, it connects the tip
lead to ground. The FXO detects this
occurrence through its tip ground detection
circuitry. In response, it connects the tip lead
to the ring lead, which completes the call.
Now, the circuit looks exactly like a loop start
circuit in the active call state.
The ground-start process is essentially a
physical layer acknowledgment protocol. The
circuit is not seized until both sides say so.
If the FXO wants to seize the circuit for an

outbound call, it connects the ring lead to
ground. The FXS detects current flow on the
ring lead from battery at the FXS to ground
at the FXO. It acknowledges the FXO's
request for service by connecting the tip
lead to ground.
The FXO detects this connection through its
tip ground-detection circuitry. In response,
it connects the tip lead to the ring lead,
which completes the call. Now the circuit
looks exactly like a loop-start circuit
in the
When Sending A DialTone

LoopStart
GroundStart
When Ringing
Loop Start
Ground Start
E&M
E&M trunk lines commonly
connect a small branch office
PBX to a PBX at the
headquarters. E&M lines are
used between PBXs, because
they have the ability to forward
the called number automatically
from one switch to another.
You can also apply the Erlang
formula to determine how many
E&M trunk lines are needed
between the two offices. Each
E&M line carries exactly one call
at a time, so four E&M lines
may carry four concurrent calls
from the branch to
headquarters.
T1 CAS SF Format
Time slot
8 bits
24 * 8 bits + 1 bit = 1 frame (193 bits)
1
10 11 12 13 14 15 16 17 18 19 20 21 22 23 24
1 bit
sync.
12 frames = Super Frame

1
10
24 * (7 bits + 1 robbed bit) + 1 bit = 1 frame (193 bits)
Time slot
7 bits +
1 robbed bit
11
12
E1 R2 CAS (Cont.)
Time Slot 1
Time slot 17
Frame 1 Indicates start of multiframe
Frame synchronization
Frames 216 Carry signaling (ABCD bits) for two

voice channels
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32
16 frames
=
Multiframe
2.048 Mb/s
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.
15.
16.
1. Frame: Start of Multiframe

Frame: Signaling for Voice Slots
2 and 18
3 and 19
4 and 20
5 and 21
6 and 22
7 and 23
8 and 24
9 and 25
10 and 26
11 and 27
12 and 28
13 and 29
14 and 30
15 and 31
16 and 32
ISDN
D Channel 16 kb/s (Signaling)
ISDN BRI
2 B channels (Voice)

ISDN E1 PRI
30 B Channels (Voice)

ISDN T1 PRI NFAS
Configuring Voice Ports

Understanding Dial Peers
Dial Peers and Call Legs

Source
Destination
Packet
Networ
k
Call Leg 1
(POTS dial peer)
Call Leg 2
(VoIP dial peer)
Call Leg 3
(VoIP dial peer)
Call Leg 4
(POTS dial peer)
Types of Dial Peers

A dial peer is an addressable call
endpoint.
Dial peers establish logical connections,
called call legs, to complete an end-toend call.
Cisco voice-enabled routers support two
types
of dial peers:
POTS dial peers: Define the characteristics
of a traditional telephony network
connection
VoIP dial peers: Define the characteristics of
a packet network connection
Dial Peer
VoiceEnabled
Router
Telephon
y
Device
POTS
VoiceEnabled
Router
VoIP
Packet
Networ
k
End-to-End Calls
Source
R1
Packet
Network
POTS
Originating
Gateway
Call Leg 1
(POTS Dial
Peer)
R1 Inbound
Destination
R2
POTS
Terminating
Gateway
Call Leg 2
Call Leg 3
(Voice Network (Voice Network
Dial Peer)
Dial Peer)
R2 Outbound
R2 Inbound
Call Leg 4
(POTS Dial
Peer)
R2 Outbound
POTS Dial Peers

Dial Peer 1
Voice Port
1/0/0
Router 1
Extention 7777
Configuration for Dial Peer 1 on Router 1:
Router# configure terminal
Router(config)# dial-peer voice 1 pots
Router(config-dialpeer)# destination-pattern 7777
Router(config-dialpeer)# port 1/0/0
Router(config-Dialpeer)# end
Configuring POTS Dial Peers Practice
3111
R1: 10.1.1.1
R2: 10.1.1.2
1/0/0
2222
1/0/0
IP WAN
1/1/0
2/1/0
PSTN
1/0/1
3112
1/1/0
3113
VoIP Dial Peers

R2(config)# dial-peer voice 2 pots
R2(config-dial-peer)# destination pattern 8
R2(config-dial-peer)# forward-digits all
R2(config-dial-peer)# port 1/0/0
Extension 7777 is calling

8888
R1
R2
IP Cloud
Extension 7777
1/0/0
L0:
10.18.0.1
R1(config)# dial-peer voice 2 voip

R1 (config-dial-peer)# destination pattern 8
R1(config-dial-peer)# session target
ipv4:10.18.0.1
PBX
Extension 8888
Configuring VoIP Dial Peers Practice
R1:
10.1.1.1
R2:
10.1.1.2
1/1/0
2/1/0
1/0/0
3111
1/0/1
3112
1/1/0
3113
1/0/0
2222
PSTN
Common Destination-Pattern
Options
+
string
(Optional) Character indicating an E.164 standard number.
Series of digits that specify a pattern for the E.164 or private dialing plan
telephone number. Valid entries are the digits 0 through 9, the letters
A through D, and the following special characters:
The asterisk (*) and pound sign (#) that appear on standard touch-tone dial
pads.
Comma (,), which inserts a pause between digits.
Period (.), which matches any entered digit (this character is used as a
wildcard).
Percent sign (%), which indicates that the preceding digit occurred zero or
more times; similar to the wildcard usage.
Plus sign (+), which indicates that the preceding digit occurred one or more
times.
Circumflex (^), which indicates a match to the beginning of the string.
Dollar sign ($), which matches the null string at the end of the input string.
Backslash symbol (\), which is followed by a single character, and matches
that character. Can be used with a single character with no other
significance (matching that character).
Question mark (?), which indicates that the preceding digit occurred either
zero times or one time.
Brackets ([ ]) indicate a range.
Parentheses (( )), which indicate a pattern.
(Optional) Control character indicating that the value is a variable-length dial
string. Using this control character enables the router to wait until all digits are
received before routing the call.
Matching Inbound Dial Peers

Configurable parameters used for matching
inbound dial peers:
incoming called-number: Defines the called
number or DNIS string
answer-address: Defines the originating calling
number or ANI string
destination-pattern: Uses the calling number
(originating or ANI string) to match the incoming
call leg to an inbound dial peer
Port: Attempts to match the configured dial-peer
port to the voice port that is associated with the
incoming call (POTS dial peers only)
Default Dial Peer 0

Dial Peer 2
Dial Peer 1
10.18.0.
1
1/0/0
Extension 7777
IP Cloud
R1
1/1/0
R2
dial-peer voice 1 pots

destination 7777
port 1/0/0
Extension
888
dial-peer voice 3 pots

destination 8888
port 1/1/0
Dial-peer voice 2 voip

destination-pattern 8888
session target ipv4:10.18.0.1
When extension 7777 calls extension 8888, there is no dial peer on
router 2 with destination pattern 7777 to match the incoming call leg.
Router 2 matches the default dial peer 0.
Matching Outbound Dial

Peers
PLAR calls
PLAR calls automatically
connect one telephone to a
second telephone when the
first telephone goes off hook.
When this connection occurs,
the user does not get a dial
tone because the voiceenabled port for that
telephone is preconfigured
with a specific number to
dial. A PLAR connection can
work between any type of
signaling, including recEive
and transMit (E&M), FXO, or
FXS, or any combination of
analog and digital interfaces.
PLAR Calls
PBX
Ring!
555-0199
Voice Port
Configured
to Dial:
555-0199
IP WAN
Gateway
Gateway

Ccna Voice

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.

CCNA Voice Introduction

Cisco Unified Communications Manager

Media Exchange (RTP)

Gateway Hardware Platforms (Cont.)

Cisco VG224 and VG248 Gateways

Cisco AS5850 Gateway

Cisco 827-4V Router

Cisco ATA 186

Cisco 7200 Series Routers

Cisco 2800 Series Routers

Cisco 3800 Series Routers

Cisco Catalyst 6500 Series

Cisco Catalyst Family of

Cisco Catalyst 6500

Cisco Catalyst 4500

Cisco Catalyst 3750

Cisco EtherSwitch Network Module

Three Ways to Power Cisco IP

Two Types of PoE Delivery

IEEE 802.3af Power over Ethernet:

Cisco Prestandard Device

IEEE 802.3af Device

IEEE 802.3af PSE

Detect Voltage Pin6

Broadest Choices Among IP

Cisco Unified Communications

Entry-Level Cisco IP Phones

Basic-featured Cisco IP phones for low-to-medium

Midrange Cisco IP Phones

Cisco Unified IP Phone

Full-featured Cisco IP phones for medium-to-high telephone use

Upper-End Cisco IP Phones

Addresses needs of executives

Other Cisco IP Phones

Cisco 7905 IP Phone

Third-Party H.323 Endpoints

Third-Party SIP Phones

Cisco 7960 IP Phone

Third-Party SIP Endpoints

Cisco IP Phone Startup Process

Cisco IP phone obtains power from the switch

Contains: option 150 or option 66

If the FXS wants to seize the

When the FXO is a telephone,

If the FXS side wants to seize a ground-start

If the FXO wants to seize the circuit for an

When Sending A DialTone

12 frames = Super Frame

24 * (7 bits + 1 robbed bit) + 1 bit = 1 frame (193 bits)

Frames 216 Carry signaling (ABCD bits) for two

1. Frame: Start of Multiframe

D Channel 64 kb/s (Signaling)

D Channel 64 kb/s (Signaling)

Configuring Voice Ports

Dial Peers and Call Legs

Types of Dial Peers

POTS Dial Peers

Configuring POTS Dial Peers Practice

VoIP Dial Peers

Extension 7777 is calling

R1(config)# dial-peer voice 2 voip

Configuring VoIP Dial Peers Practice