CCNA Voice

* The world at the end of the 20th Centure:
Voice Video
Data
-Three separate networks: Voice, Data, Video.

-Difficult to integrate applications.
- Each area is its own world: Infrastructure, staff…………….
* Cisco Goals: Unified Communications:
Voice Video
Data
- B.W capabilities have been increasing for years.

- Service Providers see the opportunity.
- Opportunity now come to business and homes.
* Why an organization would use VOIP:
Trackable cost savings:

- Moves, add and changes (MACs).
- Reduce wiring.
- Reduce telecommuter and branch office expenses.
- IT staff consolidation.
- Application consolidation.
- Toll bypass.
Soft cost savings:
- Single inbox for messages ( Voice mail, Fax, Email).
- Extension mobility ( save office space).
- Internet website integration (happy customer).
- Open architecture ( multi-vendor solution).
* The old to the new:
* How To Get The most From The Series:
-Repetition, Repetition, Repetition.
-Take note, Write down key information you hear.
-Purchase a lab router:

- Cisco 2801 with CUE AIM module, 2 FXS, 2FXO
- A couple Cisco IP Phone.
- CUCM software
- GNS3
-Dig deeper.
- Fall in love.
The Cisco Voice Structure Model:
Eng: Mohamed Saied Afify

Call Processing:
-Cisco Unified Communication 500 ( UC500).
-Cisco Unified Communication Manager Express (CME).
-Cisco Unified Communication Manager Business Editions.
- Cisco Unified Communication Manager (CUCM).

Call Processing:
1-Cisco Unified Communication 500 ( UC500):
- Support for 8 to 48 IP Phones.
- Built-in eight-port inline power switch.
- External Music on Hold port (MoH) .
- (FXO) ports (for analog PSTN connections).
- (FXS) ports (for analog phone/modem/fax connections).
- Routing / (NAT) / Firewall Support .
- (VPN) support for up to 10 users.
-Optional integrated 802.11 wireless network connectivity.

- Integrated voice mail and Auto - attendent.
Call Processing:
2-Cisco Unified Communication Manager Express:
- CME is the next step up from UC500.
-Support amax of 450 IP Phone.
-Target market: Enterprise Branch office / SMB.
-Voice mail Support added

through Unity Express (CUE).
-Runs on Cisco ISRs

(2800, 2900, 3800,…….).
- Support command line and

Cisco Configuration Professional
(CCP).

Call Processing:
3-Cisco Unified Communication Manager Business Editions:
-Provide scalability to 500 IP Phone.
-Combine three application in one solutiion:
- Cisco Communication Manager.
- Cisco Unity Connection.
- Cisco Unified Mobility.
- Great……But not Redundancy. MCS 7835

Call Processing:
4-Cisco Unified Communication Manager:
-Scale to 30,000 IP Phones per cluster
-Multi – Server Redundancy.
-Multi – Site support.
- Show me the money !!!!!!!!!!!!

* Cisco Call Processing Platforms:

Applications:
- Cisco Unity ( Voice mail).
- Cisco Unity Express.

- Cisco Unity Connection.
- Cisco Unity.
- Cisco Unified Presence.
- Additional applications.

Applications:
1-Cisco Unity Products:
A-Cisco Unity Express:
- Up to 300 mailboxes.
-The smallest unity solution you can

add to your network.
- AIM and NM form factors.
-Linux Based, Voicemail from Email,

Web, Phone.
- Basic Interactive Voice Response (IVR).

Applications:
1-Cisco Unity Products:
B- Cisco Unity Connection:
- Linux – Based appliance ( same as CUCM).
- Currently 20,000 max mailboxes per server.
- Access Voicemails from anywhere.
- LDAP directory server integration.
- MicroSoft Exchange support.

Applications:
1- Cisco Unity Products:
C-Cisco Unity :
- Run on a windows platform, full integrates in to MS Exchange,

Lotus Notes, Novell Group Wise e-mail systems.
- Support fully unified messaging.
- Currently 15,000 max mailboxes per server.
- Painful to setup.
- Still has unique features, but fading quickly.

Applications:
2- Cisco Unified Presence:
-Provides status information.
- Integration with Cisco Unified Personal Communicator (CUPC).
-Enterprise IM:
(CUP incorporates the Japper Extensible Communication Platform (XCP))
- Message compliance.
- Secure messaging.

Additional applications:
- IVR / Auto attendant
- Cisco Unified meeting place.
- Cisco Unified Mobility.
- Cisco Emergency Responder

End Points:
- IP Phone.
- Entry level IP Phone

- Business class IP Phone
- Touch screen IP Phone
- Specialty devices
- Wireless / Cell Phone.
- Video Phone.
- IM Client.

Cisco Unified
1- Entry-level IP phones IP Phone 7911
The following IP phones fall under this category:
- Cisco 3911, Cisco 7906G, 7911G and Cisco 7931G

- Basic-featured Cisco IP phones for low-to-medium
telephone use
- Single line
- Inline power support
- Basic XML service support
Cisco7931

2-Business-class IP phones:
-These phones all support two or more lines

- larger displays, high resolution, full colour display
- full-duplex speakerphones
- Inline power capabilities.
Example: Cisco 7941, 7942,7945 G

Cisco 7960, 7961,7962,7965( six lines)

3-Touchscreen Cisco IP Phones
– Cisco 797X series
– Eight line phones
– Full color
– Touch screen display
– 7975G ( display 5.6 inch)

4-Specialty device:
Cisco7985 G Cisco 7937G Conference Station
Cisco 7937G Conference Station

-Cisco ATA 186/188
The Cisco ATA boxes are small devices that can convert up to two analog
phones (per ATA) into VoIP devices.
This can be the ideal solution to bring legacy fax machines and all-in-one
devices into the VoIP network.
- The ATA 186 provides a single Ethernet interface.

Cisco IP Communicator

Cisco 7914/7915/7916 Expansion Modules

* Understanding Analog Connectivity.
- What is analog connectivity?
- Electric waveforms: more than you ever knew.
- Understanding analog signaling.
* Understanding Digital Connectivity.
- Problem with analog connections.
- Converting Analog to Digital Signal.
* Solving the wiring problem: TDM
- Understanding T1 and E1 CAS specifics.
- Understanding T1 and E1 CCS specifics.

* What is analog connectivity?
-Analog transmission: Using some property of the transmission media to convey

a signal.
- Thomas Edison’s phonograph in 1877.

- Record players.
- Braille for the blind.
- Typical home telephone lines.
- Analog phone lines use the properties of electricity for voice transmission.

* Properties of Electricity:
-As you speak into analog phone your voice is converted in to electricity.
- The properties of the electricity are used to convey the properties of your voice

* Analog Signaling: 1- Loop and Ground start:
a- Loop Start:
When the receiver is on-hook, the circuit is broken.

b- Ground Start:
Off-hook signal accomplished by temporarily grounding the ring wire.

Analog Signaling:
2- Supervisory Signal:
-Used to send the following signals over an analog line:
- On-hook
- Off-hook
- Ringing
- Ringing is sent using AC current rather than DC.
3-Informational Signal:
-Used to send the following signals over an analog line:
- dial tone - Busy - Ring back
- Congestion - No such number - Receiver off-hook

Analog Signaling:
4- Address Signal:
- Used to send the dialing information over an analog line:
1- Pulse
2- Dual Tone Multi frequency (DTMF)

* Problems with analog connections:
1- Distance Limitation:
2- Wiring Requirements:

How to turn spoken voice into Bits?
The 4 step recipe:
1- Take many samples of the analog signal.
2- Calculate a number representing each samples (AKA Quantization).
3- Convert that number to binary.
4- (Optional) Compress the signal

Converting Analog to Digital Connections:
-Digitizing Voice:
Step 1: Sample the signal
-The famed Dr: Nyquist formula
- If you sample at twice the highest frequency, you can accurately
reconstruct a signal digitally.
Common Frequencies:
- Human ear 20 - 20,000 Hz

- Human speech 200- 9,000 Hz
- Telephone channels 300-3,400 Hz
- The Nyquist theorem 300-4,000 Hz

Step 2: Perform Quantization on the sample:
Known as Pulse Amplitude Modulation (PAM)

Step 3: Convert the (Quantized) signal to binary:
=10011001
2 PCM method: A-Law, µ Law
- A Law makes more sense
Know as Pulse Code Modulation (PCM)

Step 4: Optionally Compress The signal:
You can :
1- Send all these samples
2- Send just the changes
3- Build a codebook
* Standard voice sample: 64 Kbps
* Common compressed value: 8 Kbps (G.729)

* Moving from Analog to Digital:
Each number represents a sound that some one made while speaking in to
telephone handset

Time Division Multiplexing Voice Channels:

The problem of signaling :
* In our analog world, signals = Electrical frequencies
* Once voice is digitized, signals must be 1s, 0s.
* Two methods of handle signaling in the digital realm:
1- Channel Associated Signaling. (CAS)
2- Common Channel Signaling. (CCS)

1- Channel Associated Signaling: (CAS)
-The least significant bit in every 6th frame is signaling.
-T1 CAS steals bits from voice channel to transfer signaling information. Called (RBS)

2- Common Channel Signaling: (CCS)
- CCS dedicates a signaling channel in T1 and E1 lines.
- Most common signaling protocol is Q.931. ( used for ISDN circuit)
- Most popular connection between voice system worldwide.
- More flexibility with signaling message.

- Higher security.
- More B.W for the voice bearer channels.
- CCS also allows PBX vendors to communicate proprietary messages (and features) between
their PBX systems using ISDN signaling, whereas CAS does not offer any of these capabilities.
-When using CCS configurations with T1 lines, the 24th time slot is always
the signaling channel.
- When using CCS configurations with E1 lines, the 17th time slot is always
the signaling channel.
- Other signaling protocols used with CCS configuration is SS7 ( communicate between COs)

* Pieces of the PSTN.
* Difference between PBX and Key systems.
* Connection to and between the PSTN.
* PSTN Numbering Plans.
* Understanding Voice Codecs.
* Roles of Digital Signal Processor.
* Choosing Acodec and Sample Size.
* VOIP B.W Saving measures.

* Pieces of the PSTN:
1- Analog Telephone
2- local Loop
3- Co Switch
4- Trunk
5- private Switch
6- Digital Telephone

* Different between PBXs and Key Systems:
-PBX Systems:
- Typically have digital PSTN connections.
- Provide each users a unique extension.
- Support a large number of feature.
-Key Systems:
- Typically have analog PSTN connections.
- Uses shared lines between phone.
- Support a smaller number of feature.

* Connection to and between PSTN:

* PSTN Numbering Plans:
- PSTN Numbering Plans managed under the ITU (E.164) Standard.

- Country Code.
- National destination code.
- Subscriber number.
-Ex: North American Numbering Plan (NANP)

- Country Code.
- Area Code.
- Central office code.
- Station code.

* Understanding Voice Codecs:
- The powers that be created a measurement system known as a
Mean Opinion Score (MOS) to rate the quality of the various voice codecs.
- The MOS is expressed as a single number in the range 1 to 5, where 1 is lowest
perceived audio quality, and 5 is the highest perceived audio quality measurement.

* Table shows how each audio codec fared in MOS testing
-You will hear two codecs continually repeated: G.711, G.729.
- G.729 comes in two different variants: G.729a, G.729b

* Roles of Digital Signal Processor:
- Cisco designed its routers with one primary purpose in mind: routing.
- In the realm of VOIP, the network requires the router to convert the loads of voice
in to digitized, packetized transmission.
-This task would overwhelm the resources you have on the router.
- DSPs offload the processing responsibility for voice related task from the processor
of the router.
- DSP is a chip that performs all the sampling, encoding, and compression functions
on audio coming into your router.

-Based on the DSP requirements given by the Cisco DSP calculator, you can
purchase one or more of the following PVDM:
■ PVDM2-8 : Provides .5 DSP chip
■ PVDM2-16: Provides 1 DSP chip
■ PVDM2-32: Provides 2 DSP chips

* Choosing A codec and sample size:
- G.711 G.729 G. 726 GSM FR
G.722 G.723 G.728
-Sample size dictates the a mount of audio included in each packet (default 20 ms)
- Larger samples = B.W saving
- Larger samples= more delay
- Bytes per sample = (Sample size * Codec B.W) / 8
Ex: ??? = ( 0.02 * 64000 )/8
= 160

* Adding in data link and network overhead:
- Ethernet 18 bytes
- Frame relay 4-6 bytes Layer 2
- PPP/ MLPPP 6 bytes
- IP 20 bytes
- UDP 8 bytes Network + Transport layer = 40
- RTP 12 bytes
160 + 18 + 40 = 218 Bytes/ Packet

* Tunneling: Bonus overhead
- GRE/ L2TP 24 bytes
- MPLS 4 bytes
- IPsec 50-57 bytes
* Adding it all together:
-Total B.W = Packet Size * Packet per second
?? = 218 * 50
= 87200 bps
= 87.2 kbps

* VOIP B.W saving measures:
1- Voice Activity Detection (VAD): Suppress silence in the conversation
Average 35% B.w saving
2- Compressed RTP : Compresses Network and Transport layer header from
40 bytes to 2-4 bytes
B.w saving codec dependent ( a round 40 % with G.729 codec)

* Modern Voice: VOIP Foundations
- Call Processing model.
- Deployment model.
* Cisco Voice Gateway Types

- Analog Voice Interfaces (FXS-FXO-E&M).
- Digital Voice Interfaces.
* Voice Gateway Communication Protocol

- H.323 - SIP - MGCP - SCCP.
* Voice and Video Transport Protocol

- RTP -RTCP.
* Call Control Models: (Distributed)

* Call Control Models: (Centralized)

* Campus IPT Design:

* Central Site Deployment Design:

* Distributed, Multi cluster Design:

* Cisco Voice Gateway Types:
- Analog Voice gateway ( one call per port).
- Digital Voice gateway ( Multiple call per port).

1- Analog Voice Ports: FXS

2- Analog Voice Ports: FXO

3- Analog Voice Ports: E&M

* CME voice gateway supporting analog voice connections:

* Digital Voice Ports:
-T1 and E1 CAS
-T1 and E1 CCS ( Also called PRI)
- BRI ( Basic Rate Interface)

ISDN
D Channel 16 kb/s (Signaling)
ISDN BRI 2 B channels (Voice)
ISDN E1 PRI 30 B Channels (Voice)
ISDN T1 PRI NFAS 23 B Channels (Voice)

24 B Channels (Voice)
* Connecting CME to other Voice Systems:

* Voice Gateway Communication Protocol:
PSTN
PSTN

- The Unified Communication System supports four methods of Gateway
Communication:
1- H.323
2- MGCP
3- SIP
4- SCCP

* Cisco Unified Communications Manager Signaling and Media
Paths:
Cisco Unified Communications Manager Express
(SCCP / SIP) (SCCP / SIP)
Media Exchange — (RTP)
- Cisco Unified Communications Manager performs call setup using a Signaling Protocol
(SCCP/SIP).
- Media exchange occurs directly between endpoints using RTP.

- Voice and Video Transport Protocols:
1- RTP
2- RTCP

* Beginning With Vision.
* Three key roles of catalyst switches.
* Connecting and Powering Cisco IP Phones.
* VLAN Concepts and Configuration.
* Understanding Voice VLANS.
* Understanding the Cisco IP phone Boot Process.

* Beginning With Vision:

* Three key roles of catalyst switches:
1- Inline power / Power Over Ethernet .
2- Dual VLANs / Voice VLANs / Auxiliary VLANs.
3- Class Of Services / Quality Of Services.

* Connecting and Powering Cisco IP Phones:
- RS 232
- 10/100 SW
- 10/100 PC

-There are three ways of providing power to your phone:
- Power Brick (Wall Power).
- Midspan Power (Powered Patch Panel/ Power injector).
- Power over Ethernet (POE) Switch.

1- Power Brick (Wall Power):
Ethernet
Connection
Power Brick
Network
Non-POE
Switch

2- Powered Patch Panel / Power injector:
Power and
Ethernet
Provides Power to
entire patch panel Ethernet
Only
Network
Non-POE
Switch

3- Power over Ethernet (POE) Switch:
Power and
- Cisco Inline Power Ethernet
Network
- Cisco IEEE 802.3af
POE Switch

* IEEE 802.3af Classifications:

* 802.3af Power Classes:

* VLAN Concepts and Configuration:
- The Normal Switching World:

A usual Switch
- One collision domain per port.
- Broad cast sent to all port
- One subnet per LAN.
- Very limited access control. PC 1 PC 2

- There are many benefits to using VLANs in an organization:
- Logically group users.
- Segments broad cast domain.
- Subnet correlation.
- Access control.
- Quality of Service.

* VLAN Trunking / Tagging:

* Understanding Voice VLANS:

* Understanding the Cisco IP phone Boot Process:
1- Cisco Switch detects POE Capabilities.
2- Switch sends Voice VLAN via CDP.
3- IP phone receives DHCP request including

option 150.
4- IP phone contacts TFTP server, receives

configuration file.
5- IP phone registers with CME.

*Preparation the infrastructure for VOIP:
- Beginning With Vision.
- Configuring IP Phone DHCP support.
- Configuring NTP.

Voice 172.16.1.0
Data 172.16.2.0
* Configuring a Trunk to the CME Router:

* Configuring Inter-VLAN Routing:

* Configuring DHCP Services on a router:
1- Exclude any necessary IP addresses.
2- Create DHCP pool.

- Define Network.
- Define default router.
- Define DNS setting.
- Define any other option (150).
3- Configure IP helper- address, if necessary


* Configuring a Router-Based DHCP Server:
.10
.10

* Setting the Clock of a Cisco Device with NTP:
- It allows Cisco IP Phones to display the correct date and time to your
users.
- It assigns the correct date and time to voicemail tags.
- It gives accurate times on Call Detail Records (CDR), which are used
to track calls on the network.
-It plays an integral part in multiple security features on all Cisco

devices.
-It tags logged messages on routers and switches with accurate time
information.


* Configuring NTP:
-Configure NTP Server.
- Optionally, configure one or more of your devices as NTP master.

* Licensing and Model for Cisco Unified CME.
* Installing Unified CME on Cisco router.

* CME……… It is not free:
1- IOS license.
. Licenses the IOS software on the router.
2- Feature license.
. License CME for specific number of users.
3- Phone user license.

. Licenses the IP Phone to interact with CME or CCM.

Cisco Unified Communications Manager Express Feature Licenses

ISR bundle examples

* CME design model / configuration model:
1- PBX model.
2- Keyswitch model.
3- Hybrid model.

1- PBX model :

2- Key switch model :

3- Hybrid model :

* Getting the necessary files:
1- Basic files.
2- GUI files.
3- XML Templates files.
4- MOH files.
5- Script files.
6- Miscellaneous files.

* Installing CME:
1- Get the necessary files from the Cisco website.
2- Place the files on an accessible TFTP server.
3- Copy the files to the router flash memory.
- Using the ( Copy ) command ( hourly).
- Using the ( archive ) command (salary).

Getting Familiar with CME administration

* CME administration options:
- Command line.
- Graphic User Interface (GUI).

* Managing CME using the command line:
- More flexible.
- Support all CME feature.
-To access the command-line interface (CLI) of the CME router, use
one of three methods:
1- Console Port.
2- Telnet access.
3- SSH access

-Telephony – service.
- ephone.
- ephone-dn.
- Dial-peer.

* Managing CME using GUI:
1- Integrated CME GUI.
2- Cisco Configuration Professional (CCP).
- CCP Express.
- CCP.

* Integrated CME GUI

* Creating a CCP Community of Managed Devices

* Connecting to the CME Router Securely

* Unified Communications Initial Configuration

1234

* Ensuring the foundation.
* CME: Three Key commands.
* Specifying the necessary load information.

* Ensuring the foundation:
1- Know the IP Phone boot process!
2- Voice Vlans configured, CDP enabled.
3- DHCP services configured, option 150 enabled.
4- TFTP server configured with correct files.
….Phones are now coming in droves to CME….

* CME: Three Key commands
Issued from Telephony – Service configuration mode
- Max-DN.
- Max-Ephones.
- IP Source – Address.

* Specifying the necessary load information:
- CME extracts firmware (load files) into flash for supported devices.
- Newer CME versions organize these into sub directories.
- Firmware must be specified using the (load) command under
Telephony-Service mode.
- These file should also be made available via TFTP.

Ephones and Ephone-DNs
Part (2)
* Understanding the purpose.
*Configuring Ephone – DNs.
* Configuring Ephone.
* Possibilities with EPhone-DNs.
* Shared line configuration.
* Auto- Registration.

- Understanding the purpose.
- Configuring Ephone – DNs.
- Configuring Ephone.

* Understanding Ephone-DNs:
- Ephone - DNs are representations of directory numbers.
- Can be Single-Line or Dual-Line.
Single-Line Dual-Line
. Only able to handle asingle . Handles two simultaneous Calls.

active call at atime.
. If on active call, incoming call . Necessary for call waiting, Conference

receive busy signal. calling, Consultative transfers.

* Understanding Ephone:
- Ephone are representations of Cisco IP Phones.
- Are linked to the device by the mac address
- Printed on the box of the Cisco IP Phone.
- Printed on the back of the Cisco IP Phone.
- From the setting > network configuration menu of the IP Phone.

* Configuring Ephone and Ephone –DNs:
- Create necessary Ephone – DNs.
- Create necessary Ephone.
- Associate Ephones and Ephone –DNs using the Mystical Button Command.

- Possibilities with EPhone-DNs.
- Shared line configuration.
- Auto- Registration.

* Understanding Shared Lines and Button overlay:
- The following creates a shared line configuration:
- Problem: One call per Ephone –DNs, one active user.
1010 1010
Ephone 8 Ephone 9


* Understanding Auto- registration and assignment:
- By default, CME registers and Phone ( you can disable this).
- Ephone registration are not saved.
- Auto- assignment associates Ephone-DNs to new Ephones.
- Auto- registered phones need to be restarted to auto assign.

Cisco CME: Voice Productivity Features Part (1)

- Configuring a Voice Network directory.
- Configuring Call forward.
- Configuring Call Transfer.
- Configuring Call Park.
- Configuring Call Pickup.

* Configuring Network directory:

* Configuring Call Forward:

5

* Configuring Call Transfer:

* Configuring Call Park:
- Allow you to “Park” a call on hold.
- Configured using the “park slot” command, which has many options
- Reserved – for DN.

- Time out Seconds.
- Limit Count.
- Notify DN.
- Only.
- Recall.
- Transfer DN.
- Alternate DN.
- Retry Seconds.

* Configuring Call Pickup:
- Answer another ringing phone in the network.
-Three types of call Pickup.
- Directed Pickup.
- Local group Pickup.
- Other group Pickup.
Group 1101 Group 1102

1101
1101
1101
1102
1102
1102
1001 1004
1003 1006
1002 1005
Group 1101 Group 1102

Cisco CME: Voice Productivity Features Part (2)

- Configuring Intercom.
- Configuring Paging.
- Configuring After-Hours Call Blocking.
-Configuring Music on Hold.(MOH)

* Configuring Intercom:
30
31
3
4 : 30
A101
A100
5
2 : 31
Assistant Manager

There are three other arguments you can use with the intercom
command to tune the functionality:
■ barge-in: Automatically places an existing call on hold and causes the intercom
to immediately answer.
■ no-auto-answer: Causes the phone to ring rather than auto-answer on

speakerphone.
■ no-mute: Causes the intercom to answer with unmuted speakerphone rather

than muted. Although this is beneficial to allow immediate two-way
conversation, you run the risk of one side barging into existing
conversations or background noise.

* Configuring Paging:
1- Paging is a one – way speaker phone – based announcement.
2- Accomplished by creating a paging number and assigning

IP Phones to the paging number.
3- Each IP Phone can only be assigned one paging number.
4- Supports unicast or multicast mode.
5- Supports multiple paging groups.
Paging Group 5510 Paging Group 5511

- Configuring Unicast, Single-Group Paging
32
5510
32
2
32
-To multicast paging, you could modify the paging command

with the following syntax:
32

- Configuring Multiple-Group Paging
32
5510
33
5511
34
5512
32,33
32
32
33
33
* Configuring After-Hours Call Blocking:
After-hours call blocking has three major steps of configuration:
Step 1: Define days and/or hours of the day that your company
considers off-hours.
Step 2: Specify patterns that you want to block during the times
specified in Step 1.
Step 3: Create exemptions to the policy, if needed.

- Configuring After-Hours Time Ranges and Dates

- Configuring After-Hours Block Patterns

- Configuring After-Hours Exemptions

* Configuring Music on Hold:
- Allow you to stream a .Wav or .au file stored in router’s flash memory.
- Supports unicast or multicast mode.
- Don’t get caught playing copyrighted stuff !

Understanding The CME Dial - Plan ( Part 1)

- Configuring Physical Voice Port Characteristics.
- Configuring a T1 CAS PSTN Interface.
- Configuring a T1 CCS PSTN Interface.
- Understanding Dial Peers.
-Understanding Call legs.
- Configuring POTS and VOIP Dial-Peers.
- Dial-Peer wildcards and digit manipulation.
- Configuring a North American PSTN Dial-Plan.
- Understanding outbound and Inbound Matching

* Configuring Physical Voice Port Characteristics:
- Configuring Analog Voice Ports:
- Foreign Exchange Station Ports (FXS)
- Foreign Exchange Office Ports (FXO)
- Configuring Digital Voice Ports

FXS ports have three common areas of configuration:
- Signaling.
- Call progress tones.
- Caller ID information.

2621 XM
NM-HD-2VE
module 1 module 0
VWIC-IMFTT1 VIC-2fx Fa 0/1 Fa 0/0
1 0
1/0/1

Foreign Exchange Office Ports
Two additional commands are of note:
- dial-type
- ring number

* Configuring a T1 CAS PSTN Interface:

* Configuring a T1 CCS PSTN Interface:

* Understanding Dial Peers:
-Types of Dial Peers
1- POTS Dial Peers
- Connect to any traditional telephony

network or devices.
-Define number(s) reachable through

a given port.
2- VOIP Dial Peers
- Connect across any packet-based network.
- Define number(s) reachable through at a given IP address.

* Understanding Call legs:
- Call legs define the voice route path.
- Every change in network (type) requires anew call leg.
- Call legs are define inbound and out bound.

* Configuring POTS Dial Peers:
Router B
3301
3301
3301
1/0/0
3302
3302
1/0/1 Eng: Mohamed Saied Afify
* Wildcards You Can Use with the destination-pattern Command

* Configuring VOIP Dial Peers:
330
330.
10
10..

* Configuring a North American PSTN Dial-Plan:

* Understanding outbound and Inbound Matching:

- Outbound Dial Peers:
The most specific destination pattern always wins

Understanding The CME Dial - Plan ( Part 2)

- Understanding outbound and Inbound Matching.
- Digit Manipulation.
- Understanding and Implementing CME Class of Restriction.

* Understanding outbound and Inbound Matching:

- Outbound Dial Peers:
The most specific destination pattern always wins

* Inbound Dial Peer:
1- Match the dialed number (DNIS) using the incoming called-number dial
peer configuration command.
2- Match the caller ID information (ANI) using the answer-address dial

3- Match the caller ID information (ANI) using the destination-pattern dial

4- Match an incoming POTS dial peer by using the port dial-peer

configuration command.
5- If no match has been found using the previous four methods, use dial peer 0.

* Inbound Dial Peer Matching Examples:

Dial Peer 0
- Any Voice codec.

- No DID Support.
- No IVR.
- VAD enabled.
- No DTMF Relay.
- No RSVP.

* Digit Manipulation:
-The Auto –Stripping Rule of POTS Dial Peers

*Digit Manipulation Commands:
- Prefix < Digit >
- Forward Digit < Number >
- Digit Strip
- Num- exp < match > < set >
- Voice Translation – profile

Practical Scenario 1: PSTN Failover Using the prefix Command

IP WAN
PSTN

PLAR

Practical Scenario 2: Directing Operator Calls to the Receptionist
5000

Practical Scenario 3: Specific POTS Lines for Emergency Calls

* Understanding and Implementing CME Class of Restriction:
- Prevent standard employees from making international calls, but allow
management to place international calls without restriction.
- Block certain high-cost numbers (such as 1-900 numbers in the United States).
- Prevent certain internal phones from reaching executive office directory numbers.

X 3000
X 4000
X 5000

* Manually Creating A PSTN Dial Plan ( In North America)
- Emergency Calling
- 7 Digit Dialing
- 10 Digit Dialing
- 11 Digit Dialing
- International Eng: Mohamed Saied Afify

* Understanding COR:
- First, Define COR Under COR Custom
- Second, Define Outgoing COR Lists
- Third, Define Incoming COR Lists
- Fourth, Assign CORs to Dial - Peers Eng: Mohamed Saied Afify

X 3000
X 4000
X 5000

* How Routers Implement Class Of Restriction:

Ex: 2
4000
0
4000
Eme-Loca
Eme-Loca

*There are a couple important rules of COR lists that you should
know:
Rule 1: If there is no outgoing COR list applied, the call is always routed.
Rule 2: If there is no incoming COR list applied, the call is always routed.

Cisco CUCM
Administration Overview, Supporting End Device

- Cisco CUCM: Administration Overview.
- Cisco CUCM: Supporting End Device.

* Cisco CUCM: Administration Overview.
- Reviewing the place of CUCM.
- Getting familiar with CUCM Web management.
- CUCM Command line interface.

- Cisco Unified Communications Manager Administration.
- Cisco Unified Serviceability.
- Disaster Recovery System.
- Cisco Unified Operating System Administration.
- Cisco Unified Reporting.
- Command-Line Interface (CLI).

* Cisco CUCM: Supporting End Device.
- Understanding Key Device Pool Setting.
- Methods of Adding Phone to CUCM.

* The purpose of device pools:
-IP Phones have a number of criteria that must be assigned to them:
- List of CCM Servers to use.
- Codec that should be used.
- Time and date information.
- Device Pool group this configuration in to a single assignment.

* Required Device Pool Elements:
- Device Pool name.
- Cisco Call Manager group.
- Date / Time group.
- Region.
- SRST Reference.
- Softkey Template

* Methods to add Cisco IP Phones to CCM:
- Manually: Entering the MAC address and extension of each IP Phone

through the CCM admin web page.
- Automatically: CCM hands out extension numbers to newly registered

phones like DHCP addresses.
- Bulk Administration Tool: Use Excel spread sheet to generate CSV files
of devices.

Cisco CUCM: Supporting End Users

- Bulk Administration Tool.
- Locking down the Cisco IP Phone.
- Cisco CUCM: Supporting End Users.

* Methods to add Cisco IP Phones to CCM:
- Manually: Entering the MAC address and extension of each IP Phone

through the CCM admin web page.
- Automatically: CCM hands out extension numbers to newly registered

phones like DHCP addresses.
- Bulk Administration Tool: Use Excel spread sheet to generate CSV files
of devices.

* The Cisco Bulk Administration Tool:
- Help in making large additions or changes to CUCM Database:
- Phone.
- Users.
- Many (tedious) Configuration.
- Now pre- integrated in CUCM administration.
- Supports Data export and re- import.
- Exported data can also be used in place migration or data restore

( not possible with DRS).

*Understanding BAT Components:
- When making Bulk additions, two pieces need to exist:

* Locking down the Cisco IP Phone:
Variety of simple to configure security options:
- Disable PC port.
- Lock setting access.
- Gratuitous ARP protection.
- PC voice Vlan access.
- IP Phone web browser access.

Cisco CUCM: Supporting End Users.

* Benefits of CCM User Accounts:
- Users have the ability to manage their own phone.
- Most soft phone devices require user logins.
- Gives way to advanced features such as extension mobility.
- More sophisticated tracking: per user account.

* Understanding User Accounts:
- Two categories of users: End users / Application users.
- End user can be added to the CUCM database

via three main method:
- Manual.
- BAT.
- LDAP.
1- Manual Entry:
- User ID.
- Last name.
- Presence group.
- Remote destination limit.
- PIN.
- Password.

3- LDAP Lightweight Directory Access Protocol:
- LDAP is a standards-based system that allows an organization to create a single,

centralized directory information store.
- LDAP holds information about user accounts, passwords, and user privileges.
- CUCM supports LDAP integration with several widely used LDAP systems,
including the following:
- Microsoft Active Directory (2000, 2003, 2008)
- iPlanet Directory Server 5.1
- Sun ONE Directory Server (5.2, 6.x)
- CUCM can interact with LDAP in two ways:

- LDAP Synchronization
- LDAP Authentication

* LDAP Synchronization:
- When LDAP Sync is enabled, user accounts must be created and maintained in
LDAP and cannot be created or deleted in CUCM.
- Some user data (but not all) is maintained in LDAP and replicated to the
CUCM database.
- The user password must be maintained in both the LDAP system and in CUCM.
- Some user attributes are not held in LDAP and are still configured in CUCM
because those attributes exist only in the CUCM database.

* LDAP Authentication:
- LDAP Authentication redirects password authentication requests from
CUCM to the LDAP system.
- End-User account passwords are maintained in the LDAP system
and are not configured, stored, or replicated to CUCM.

* User account information:
- User account information is divided into three categories, with fields for
specific data in each category:
1. Personal and Organizational Settings:
- User ID - First, Middle, Last Name - Manager UserID
- Department - Phone Number, Mail ID
2. Password Information: Password
3. CUCM Configuration Settings:
- PIN - SIP Digest Credentials - User Groups and Roles
- Associated PCs, controlled devices and DNs
- Application and feature parameters (Extension Mobility, Presence Group, CAPF)

* LDAP Sync Mechanism:
- The first time the synchronization happens, the following events take place:
- All existing end-user accounts in the CUCM database are deactivated (not
deleted).
- Accounts whose CUCM User ID exactly matches a user in LDAP are

reactivated, and any settings from LDAP are updated or applied in the
CUCM database.
- Accounts that exist only in LDAP are created in the CUCM database.
- Any accounts that remain deactivated (meaning they do not exist in LDAP)
are deleted from the CUCM database after 24 hours.

* Configure LDAP Sync:
The basics steps to set up LDAP Sync are as follows:
1. Activate the Cisco DirSync service.
2. Configure the LDAP system.
3. Configure the LDAP directory.
4. Configure LDAP Custom Filters.

* Managing Groups, Roles and Privileges:
- Users assigned to groups.
- Groups assigned to one or more Roles.
- Roles assigned to privileges.
Users Application
Groups
Roles Privileges

Cisco CUCM
Understanding Dial Plan
Partitions and Calling Search Spaces

* Understanding Dial Plan:
- CUCM Route Architecture.
- Building Route Group.
- Building Route List.
- Building Route Pattern.

* CUCM Route Architecture:

* Call Manger Route Plan Architecture:
Route
Pattern
Route List
Route Group Route Group
Device Device Device Device

* Route Pattern Wildcards:
X : Single Digit
@ : North American Numbering Plan
! : One or More Digit
. : Access Code Termination / Digit Formatting
# : Terminates Interdigit Timeout

[xyz] [x-y] [x-yz] = Digit Set
[^ xyz] [^x-y] [^x-yz] = Negative Digit Set
Example:
x123
195xx
[139]11
[1-5]00
[13-59]1
[^1-3]11
38[^2-4]3
9011!#

* Call Manger Route Plan Architecture:
1xxx 2xxx
10.1.1.1
WAN
Route 2xxx
Pattern
10.1.1.2 PSTN
1-WAN
Route List 2-PSTN
1-WAN Route Group Route Group 2-PSTN
10.1.1.1 Device Device Device Device 10.1.1.2

- Cisco CUCM: Partitions and Calling Search Spaces
- Define Partitions and Calling Search Spaces.
- Assigning Calling privileges using Partitions and CSS.

* Partitions and Calling Search Spaces:
- Partitions: Groups of dialable numbers.
- Lines.
- Route Patterns.
- Any thing that has a number.
Internal-PT Local-PT INT-LD-PT
2001 9.[2-9]xxxxxx 9.011!

2002 9.1[2-9]xx[2-9]xxxxxx
2003
.
.
2999
- Calling Search Spaces: A list of reachable Partitions.
- assign to any “dialing” entity.
- Define calling privileges.
Internal- CSS
Out- CSS -Internal-PT
-INT-LD-PT
-Local-PT 2010

* Understanding Partitions and Calling Search Spaces.
- By default, All numbers assign to the (none) Partitions.
- By default, All devices assign to the (none) CSS.
(none) CSS
All numbers
-None
Partitions: none

* Understanding Partitions and Calling Search Spaces.
- Numbers moved into new Partition become unreachable from

the (none) CSS devices.
(none) CSS
All numbers
-None
Partitions: none
2010
EXT- CSS
9.@ - External-PT
- None
External-PT
* Practical Partitions and Calling Search Spaces.
Three types of calling restrictions should exist in your organization:
- Lobby / Public Phones: Internal extension only.
- Typical users: Internal and Local PSTN.
- Management: Internal , Local and Long distance PSTN.
Step1: Create the Partitions.
Step2: Assign number to Partitions.
Step3: Create CSSs.
Step4: Assign CSS to devices.

- Plan your Partitions a round the calling restrictions
Internal-PT Local-PT LD-PT

LD-PT
Internal-PT Local-PT
2001 9.[2-9]xxxxxx 9.1[2-9]xx[2-9]xxxxxx

2002
2003
.
.
2999
Step3,4: Create / Assign CSS:
Lobby-CSS Local-CSS Manger-CSS

-Internal-PT
-Internal-PT -Internal-PT -Local-PT
-Local -PT -LD-PT
Lobby Normal Users Manager

Phone Phone Phone

* Try it yourself:

- Plan your Partitions a round the calling restrictions
Internal-PT Manager-PT Local-LD-PT INT-PT
Internal-PT Manager-PT Local-LD-PT INT-PT
1002 1001 9.[2-9]xxxxxx 9.011!

1003 9.1[2-9]xx[2-9]xxxxxx

Step3,4: Create / Assign CSS:
Lobby-CSS Employee-CSS Manger-CSS

-Internal-PT -Internal-PT
-Internal-PT -Local-LD-PT -Local-LD-PT
-Manger-PT -Manger-PT
-INT-PT
Lobby Employee Manager

Phone Phone Phone

Cisco CUCM:CUCM Feature Overview
Understanding QOS

* Cisco CUCM:CUCM Feature Overview
- Call Park.
- Call Pickup.
- Shared Lines.
- Do Not Disturb.
- Call Back.
- Barge and Privacy.
- Services / Extension Mobility

* Quality of Service (QoS):
- Understanding the enemy.
-The three current models of QoS.
- QoS queuing method.

- Understanding the enemy:
1- Lack of B.W.
2- Packet Loss.
LAN
5 Mbps T1
3- Delay. 1.5 Mbps
- Fixed delay.
- Variable delay.
- Jitter.

* Network Requirements For Voice and Video:
1- B.W.
2- End - to – End delay. (150ms or Less)
3- Jitter. (30ms or Less)
4- Packet Loss. (1% or Less)

* The three current models of QoS:
- Best Effort.
- Integrated Services. (IntServ)
- Differentiated Services. (DiffServ)
* QoS ToolBelt:
- Classification: Involves identifying and grouping different traffic types.
- Marking: Tages or colors the packet so it can be quickly recognized elsewhere

in the network.

* Queuing Algorithms:
- Weighted Fair Queuing. (WFQ)
- Class-Based Weighted Fair Queuing. (CBWFQ)
50 Packet http
20 Packet ftp
10 Packet telnet
- Low Latency Queuing. (LLQ) (PQ-CBWFQ)

Cisco Unity Connection (CUC)
Cisco Unified Presence (CUPs)

* Overview of Cisco Unity Connection:
- CUC is one of 5 VOIP linux appliances.
- CUC integrates with legacy PBX systems via PIMG or TIMG.
- CUC integrates with CUCM using SCCP or SIP.
- CUC users ( Manually, CSV, CUCM import, or LDAP ).
- Up to 2000 mail boxes per servers.
- Access voice mail from anywhere.
- VPIM support.
- Active / Active high availability.

* How CUC Integrates with CUCM?
SCCP SIP

* How CUC Processes Calls?
- All inbound calls to CUC are handled by a series of call handlers:
1- System call handlers.
2- Directory handlers.
3- Interview handlers.
- Call Routing:
-Two primary call routing are built in CUC:
1- Direct call.
2- Forward call.

* Managing Users and Mailboxes in CUC:
- User templates make configuration easier.
- User templates basic:
1- Name.
2- Phone.
3- Location.
- COS defines many options ( timers, features, restriction).
- User creation options:
1- Manually.
2- Bulk administration.
3- Import from CUCM.
4- Import from LDAP.

Cisco Unified Presence (CUPs)

* What is the point of CUPs?
- CUCM include basic presence feature (ON/OFF hook).
- CUPs add in:
- Additional state ( Available, A way, Busy).

- Additional method ( Desk, Mobile Phone, IM, Conf).
- Calendar integration (Status Auto- Updates).
- Enterprise instant message (IM).
- Brings Cisco Unified Personal Communicator (CUPC) to the table.
- CUPC supports Desk phone, Soft phone modes.

-Desk phone mode:

- Soft phone mode:

* How CUPs / CUPC Communicate:
- Simple / XMPP- Presence information.
- SOAP- CUCM database access.
- CTIQBE – office communicator integration.
- LDAP – user database integration.
Unity

* CUCM Configuration of Unified Presence:
1- Assign user license in CUCM.
2- Associate users with phone / lines in CUCM.
3- Enable CTI on phone / line.
4- Create CSF device or UPC device in CUCM.
5- Associate user with CSF device.

* CUPs Configuration of Unified Presence:
1- Create link to voice mail server.
2- Create link to CUCM server ( CTI gate way).
3- Add LDAP server.
4- Define the CCMCIP profile.

Voice Troubleshooting

* Voice Troubleshooting: Cisco CME
- Getting in Troubleshooting groove.
- Troubleshooting registration issues.
- Troubleshooting Dial-Plan issues.

Troubleshooting: Getting in Troubleshooting groove.
- Step 1: Define the problem.
- Step 2: Gather the facts.
- Step 3: Consider the possibilities.
- Step 4: Create an action plan.
- Step 5: Implement the action plan.
- Step 6: Observe results.
- Step 7: If resolved, document.

-Troubleshooting registration issues:
1- Switch detects un powered device, Supplies power.
2- Switch provides VLAN information to the IP Phone.
3- Phone sends DHCP request, receives IP and TFTP info.
4- Phone contracts TFTP servers, retriever configuration file.
5- Phone contracts CME route listed in configuration file.

* Troubleshooting Dial-Plan issues:
- Understanding how Dial-Peers work.
- Use some key troubleshooting commands:
- Show IP interface brief.
- Show Dial- Peer voice summary.
- debug VOIP Dial- Peer.

* Voice Troubleshooting: Cisco CUCM
- CUCM common issues.
- CUCM additional troubleshooting utils.

* Troubleshooting registration using reports:
- Registration problems ……………. Where to look:
- Device configuration.
- Route plan report.
- Understanding reports:
- Finding the CUCM reports.
- CDR analysis and reporting (CAR).

* Additional CUCM utilities to save the day:
1- Cisco Real Time Monitoring Tool ( RTMT).
2- Cisco Disaster Recovery System (DRS).

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* The world at the end of the 20th Centure:

-Three separate networks: Voice, Data, Video.

- B.W capabilities have been increasing for years.

Trackable cost savings:

-Repetition, Repetition, Repetition.

-Take note, Write down key information you hear.

-Purchase a lab router:

Eng: Mohamed Saied Afify

-Cisco Unified Communication 500 ( UC500).

-Cisco Unified Communication Manager Express (CME).

-Cisco Unified Communication Manager Business Editions.

- Cisco Unified Communication Manager (CUCM).

Eng: Mohamed Saied Afify

- Built-in eight-port inline power switch.

- External Music on Hold port (MoH) .

- (FXO) ports (for analog PSTN connections).

- (FXS) ports (for analog phone/modem/fax connections).

- Routing / (NAT) / Firewall Support .

- (VPN) support for up to 10 users.

-Optional integrated 802.11 wireless network connectivity.

- CME is the next step up from UC500.

-Support amax of 450 IP Phone.

-Target market: Enterprise Branch office / SMB.

-Voice mail Support added

-Runs on Cisco ISRs

- Support command line and

Eng: Mohamed Saied Afify

-Provide scalability to 500 IP Phone.

-Combine three application in one solutiion:

- Cisco Communication Manager.

- Cisco Unity Connection.

- Cisco Unified Mobility.

- Great……But not Redundancy. MCS 7835

Eng: Mohamed Saied Afify

-Scale to 30,000 IP Phones per cluster

-Multi – Server Redundancy.

-Multi – Site support.

- Show me the money !!!!!!!!!!!!

Eng: Mohamed Saied Afify

Eng: Mohamed Saied Afify

- Cisco Unity ( Voice mail).

- Cisco Unity Express.

- Cisco Unified Presence.

Eng: Mohamed Saied Afify

-The smallest unity solution you can

- AIM and NM form factors.

-Linux Based, Voicemail from Email,

- Basic Interactive Voice Response (IVR).

Eng: Mohamed Saied Afify

- Linux – Based appliance ( same as CUCM).

- Currently 20,000 max mailboxes per server.

- Access Voicemails from anywhere.

- LDAP directory server integration.

- MicroSoft Exchange support.

Eng: Mohamed Saied Afify

- Run on a windows platform, full integrates in to MS Exchange,

- Support fully unified messaging.

- Currently 15,000 max mailboxes per server.

- Still has unique features, but fading quickly.

Eng: Mohamed Saied Afify

-Provides status information.

- Integration with Cisco Unified Personal Communicator (CUPC).

Eng: Mohamed Saied Afify