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Page 1: Ccna Voice

CCNA Voice Introduction

الرحمن الله بسم الرحيم

. أمري لي ويسر ليصدري اشرح ربقولي يفقهوا لساني من عقدة وأحلل

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التدريس هيئة أعضاء


شريف • sharif_مشرفنا soper

lumark_أستاذنا • s

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Cisco Unified Communications Manager Signaling and Media Paths

Cisco Unified Communications Manager Express

IP Phone A

Signaling Protocol


Media Exchange — (RTP)

Signaling Protocol


Cisco Unified Communications Manager performs call setup using a Signaling Protocol (SCCP/SIP).

Media exchange occurs directly between endpoints using RTP.

IP Phone B

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Gateway Hardware Platforms (Cont.)

• Special voice gateways:

Cisco VG224 and VG248 Gateways

Cisco AS5300 and AS5400 Series Gateways

Cisco 7200 Series RoutersCisco ATA 186

Cisco 827-4V RouterEOS: 05/2005EOL: 05/2010

Cisco AS5850 Gateway

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Gateway Hardware Platforms

• Modern enterprise models:

Cisco 2800 Series Routers Cisco 3800 Series Routers Cisco Catalyst 6500 Series

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Cisco Catalyst Family of Switches

Cisco Catalyst 3560

Cisco Catalyst 3750

Cisco Catalyst 6500 Cisco Catalyst 4500

Cisco EtherSwitch Network Module

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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 outlet110 V AC Wall Power to 48 V DC Converter

AC Source

No PowerPower Injector



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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.

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Cisco Prestandard Device Detection


Cisco Prestandard ImplementationPowered Device Port




It is an inline device.





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IEEE 802.3af Device Detection


IEEE 802.3af Powered Device



Detect Voltage 25K Ohm Resistor

It is an IEEE powered device.





2.8V to 10V

IEEE 802.3af PSE

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Redundancy Design Primary

Secondary orBackup

Publisher and TFTP Server (Not Req. <1000)

Publisher and TFTP Server

Publisher and TFTP Server

7500 IP phones 15,000 IP phones 30,000 IP phones

Primary 1 to 7500



1 to7500

1 to7500

15001 to22,500

7501 to15,000

7501 to15,000

22,501 to30,000

Cisco MCS 7845 Cisco MCS 7845 Cisco MCS 7845

Backups Backups

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Cisco IP Phone 7902G

Cisco IP Phone 7905G and Cisco IP Phone 7912G

Cisco IP Conference Station 7936

Cisco IP Phone 7940G

Cisco IP Phone 7960G

Cisco ATA 186/188

Cisco VG248 Analog Phone Gateway

Cisco IP Phone 7970G & 7971G

Broadest Choices Among IP Endpoints

Cisco VT Advantage

Cisco IP Communicator

Cisco WirelessIP Phone 7920

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Cisco Unified Communications Manager Endpoint Overview

Cisco Unified Communications

Manager 6.0

Cisco UnifiedIP Phones SCCP


Analog Station Gateways Third-Party

H.323 Endpoints


SCCP VideoPhones

Third-Party SIP Endpoints

Cisco SCCP-OnlyPhones

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Entry-Level Cisco IP Phones

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

Single line or directory number Message waiting indicator

Cisco Unified IP Phone 7906

Cisco Unified IP Phone 7911

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Midrange Cisco IP Phones

– 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

Cisco Unified IP Phone 794[012] Cisco Unified IP Phone 796[012]

7940 7941 7960 7961

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Upper-End Cisco IP Phones

– 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

Cisco UnifiedIP Phone 797[015]

Cisco UnifiedIP Phone 7945

Cisco UnifiedIP Phone 7965


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Other Cisco IP Phones

Cisco Unified IP Conference Station 7936

Cisco Unified IP Phone 7985

Cisco UnifiedWireless IP Phone 792[01]

Cisco Unified IP Phone 7931


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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

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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

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Cisco IP Phone Startup ProcessUnified CM Cisco TFTPDHCP



51 3


1. Cisco IP phone obtains power from the switch

2. Cisco IP phone loads locally stored image3. Switch provides VLAN information to Cisco IP phone using Cisco Discovery Protocol4. Phone sends DHCP request; receives IP information and TFTP server address5. Cisco IP phone gets configuration from TFTP server6. Cisco IP phone registers with Cisco Unified Communications Manager server7. Unified CM sends softkey template to SCCP phone using SCCP messages

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DHCP Discover

DHCP Offer

DHCP Request


Contains: option 150 or option 66

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• Firmware• SEPAAAABBBBCCCC.cnf.xml• XmlDefault.cnf.xml• SCCP-dictionary.xml• Phonemodel-dictionary.xml• Phonemodel-tones.xml

Files critical to the IP phone

TFTP Server













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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.

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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 will connect to unsuspecting outbound callers who are not the intended called party.

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Ground StartWhen 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 active call state.

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When Sending A DialToneLoopStart GroundStart

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When RingingLoop Start Ground Start

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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.

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T1 CAS SF Format

24 * 8 bits + 1 bit = 1 frame (193 bits)

12 frames = Super Frame

Time slot7 bits +

1 robbed bit

1 2 3 4 5 6 7 8 9 10 11 12

1 2 3 4 5 6 7 8 9 10 21 22 23 2411 12 13 14 15 16 17 18 19 20

Time slot8 bits

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

1 bitsync.

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E1 R2 CAS (Cont.)Time slot 17

Frame 1 Indicates start of multiframe

Frames 2–16 Carry signaling (ABCD bits) for two voice channels

Time Slot 1

Frame synchronization

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=

Multiframe2.048 Mb/s

1. Frame: Start of Multiframe 2. Frame: Signaling for Voice Slots 2 and 183. Frame: Signaling for Voice Slots 3 and 194. Frame: Signaling for Voice Slots 4 and 205. Frame: Signaling for Voice Slots 5 and 216. Frame: Signaling for Voice Slots 6 and 227. Frame: Signaling for Voice Slots 7 and 238. Frame: Signaling for Voice Slots 8 and 249. Frame: Signaling for Voice Slots 9 and 25

10. Frame: Signaling for Voice Slots 10 and 2611. Frame: Signaling for Voice Slots 11 and 2712. Frame: Signaling for Voice Slots 12 and 2813. Frame: Signaling for Voice Slots 13 and 2914. Frame: Signaling for Voice Slots 14 and 3015. Frame: Signaling for Voice Slots 15 and 3116. Frame: Signaling for Voice Slots 16 and 32

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D Channel 16 kb/s (Signaling)

2 B channels (Voice)ISDN BRI

D Channel 64 kb/s (Signaling)

30 B Channels (Voice)ISDN E1 PRI

D Channel 64 kb/s (Signaling)

24 B Channels (Voice)ISDN T1 PRI NFAS 23 B Channels (Voice)

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Configuring Voice Ports

Understanding Dial Peers

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Dial Peers and Call Legs

Packet Network

Call Leg 1(POTS dial peer)

Call Leg 4(POTS dial peer)

Call Leg 2(VoIP dial peer)

Call Leg 3(VoIP dial peer)

Source Destination

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Types of Dial Peers– A dial peer is an addressable call endpoint.– Dial peers establish logical connections, called call

legs, to complete an end-to-end 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

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Dial Peer


Telephony Device





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End-to-End Calls

Source DestinationR1 R2



Call Leg 1(POTS Dial


R1 Inbound

Call Leg 4(POTS Dial


R2 Outbound

Call Leg 2(Voice Network

Dial Peer)

R2 Outbound

Call Leg 3(Voice Network

Dial Peer)

R2 Inbound

POTS PacketNetwork


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POTS Dial Peers

Dial Peer 1

Extention 7777

Router 1

Voice Port1/0/0

Configuration for Dial Peer 1 on Router 1:

Router# configure terminalRouter(config)# dial-peer voice 1 potsRouter(config-dialpeer)# destination-pattern 7777Router(config-dialpeer)# port 1/0/0Router(config-Dialpeer)# end

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Configuring POTS Dial Peers Practice












1/1/01/1/0 2/1/0

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VoIP Dial Peers

IP Cloud

Extension 7777 Extension 8888

R1 R2


R1(config)# dial-peer voice 2 voipR1 (config-dial-peer)# destination pattern 8…R1(config-dial-peer)# session target ipv4:

R2(config)# dial-peer voice 2 potsR2(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

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Configuring VoIP Dial Peers Practice






1/1/0 2/1/0







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Common Destination-Pattern Options+ (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.

T(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.

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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)

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Default Dial Peer 0

IP Cloud

Extension 7777 Extension 888

dial-peer voice 1 potsdestination 7777port 1/0/0

Dial-peer voice 2 voipdestination-pattern 8888session target ipv4:


Dial Peer 1Dial Peer 2

dial-peer voice 3 potsdestination 8888port 1/1/0


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.

R1 R2

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Matching Outbound Dial Peers

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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 voice-enabled 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.

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PLAR Calls

555-0199 Voice Port Configured to Dial:“555-0199”


Gateway Gateway