Post on 11-Oct-2015
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transcript
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CCNA VOICE - LAB SETUP
CUCM Software
GNS3
VMWare Workstation
Call Manager Express (CME)
Cisco Unity Connection/Cisco Unity Connection Express
Cisco 2600XM, 2801 Routers (ISR/VG)
NM2V WIC Card
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CCNA VOICE INTRO
CUCM can run in VMWare
Google for GNS3 labs for CCNA Voice
Voice/Data/Video Collaboration is the key
Difficulty to integrate applications
Each area is its own world
Cisco Goal: Unified Communications
AVVID Acronym used prior to UC
Bandwidth capabilities are increasing, therefore opportunities come to businesses and
homes. ISP see this opportunity in particular
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CCNA VOICE INTRO
Why VOIP?
Cost Savings:
MAC (Moves, Adds, Changes) $70-$200 per device)
Reduced wiring
Reduced telecommuter + branch office expenses
IT Staff + Application Consolidation
Toll Bypass (Long Distance in particular)
Soft Cost Savings:
Single Inbox for messages (Voicemail/Fax/Email)
Extension Mobility (Saves Office Space)
Internet Website Integration (Happy Client )
Open Architecture (Multi Vendor Solution)
SIP is the TCP/IP of the voice world
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CCNA VOICE THE OLD TO THE NEW
Phase 1 Keep Existing PBX:
Requip existing routers for WAN + PSTN
Phase 2 Once all OK, removed PBX and use VOIP primarily.
FACT: 2 Million people in the USA still use rotary phones
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CCNA VOICE UNIFIED SOLUTIONS OVERVIEW
Core Products
Cisco Unified Communications Manager Express (Runs from router flash)
Cisco Unified Communications Manager (Runs from dedicated MCS appliance)
Cisco Unity Connection Express (Voicemail)
Cisco Unity Connection (Voicemail on dedicated MCS appliance)
Cisco Unified Presence (Tracking/IM)
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CCNA VOICE - CUCME
Max of 450 IP phones, but 100 ideally
Target Market: Enterprise Branch/Small Business Offices
Voicemail support added through CUE
Runs on Cisco ISRs (2800, 2900, 3800 etc)
Supports CLI + CCP configuration
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CCNA VOICE - MODULES
VM Module required for CUE
AIM = Circuit Board/NM = Module with handle
AIM = Flash based/NM = HDD with Linux OS
3800 Series = CLI only
CCP = Nice and exam heavy as introductory
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CCNA VOICE UNIFIED COMMUNICATIONS 500
8 to 48 phones
Integrated voicemail and auto attendant
External music on hold port
FXO/FXS Modules Analog connections
Routing and NAT features
VPN Users (10 max)
Optional 802.11 Wireless
Expensive
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CCNA VOICE CISCO UNIFIED
COMMUNICATIONS BUSINESS EDITION
500 IP Phones
CCM Communications Manager
Cisco Unity Connection
Cisco Unified Mobility
NO REDUNDANCY
FULL CUCM BUT STRIPPED TO SINGLE SERVER
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CCNA VOICE CISCO UNIFIED
COMMUNICATIONS MANAGERFULL BEAST
30,000 phones per cluster/shared database (60,000???)
Multiserver redundancy
Multisite support
Expensive
Flagship product for Cisco
CME supports up to 100 phones, CUCM exceeds 100 plus REDUNDANCY.
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CCNA VOICE CISCO UNITY CONNECTION +
UNIFIED PRESENCECisco Voicemail options:
Cisco Unity Express
Cisco Unity Connection
Cisco Unity
Cisco Unity
Original version of unified messaging
Runs on Windows OS
Exchange + Domino integration
PAINFUL to setup
Still has unique feature set, but is fading. (Direct tether to Exchange)
15000 users per server
Cisco Unity Connection
Linux Based Appliance
Previously IMAP only now Exchange integration
20000 Users per server
Unlimited number of telephone integration
Featureiffic
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CCNA VOICE CISCO UNITY CONNECTION +
UNIFIED PRESENCE CONTCisco Unity Express
Voicemail option with CME
AIM + NM form factors (ISM and SM)
250 Users Max
Basic interactive voice response (IVR)
Auto attendant, email integration (Exchange)
Cisco Presence
Provides status information
Integrates into nearly every IT facet (CUCM/IP Phones/Unity/LDAP etc..)
Uses industry standard SIP to collect data
Integration with CUPC Cisco Personal Communicator
Enterprise instant messaging
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UNDERSTANDING ANALOG CONNECTIVITY
Pulse dialing around for 40 years (Rotary phones)
What is analog connectivity?
Transmission: Using some property of the transmission media to convey a signal.
Thomas Eddisons Phonograph in 1877-1900s
Record players
Braille
Typical Home Telephones Lines
Analog phone lines use the properties of electricity for voice transmission
Phonograph: Store signals in a cylinder, bumps in tin foil (Stored and relayed
voice) magnetic fields/wave form signal.
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UNDERSTANDING ANALOG CONNECTIVITY
CONT..
Properties of electricity
VOLTAGE X AXIS vs TIME Y AXIS
As you speak into an analog phone, your voice is converted into electricity.
The properties of the electricity are used to convey the properties of your voice.
ANALOG: Loop and Ground Start
Loop Start: PHONE has 2 wires that run in a complete circuit to a battery.
PHONE ---------------------------RING WIRE--------------------------------BATTERY
PHONE ---------------------------TIP WIRE------------------------------------BATTERY (CURRENT
DETECT)
When the receiver is ON-HOOK, the circuit is broken, when the phone is OFF-
HOOK the circuit is complete. (Hence dialtone when phone is off hook)
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UNDERSTANDING ANALOG CONNECTIVITY
CONT..Ground Start
Off hook signal temporarily accomplished by grounding the RING wire.
Grab outbound line and call inbound at the same time = GLARE
PHONES ---------- PBX ---------MULTIPLE LINES-----------CENTRAL OFFICE
Shoots ground signal down ring (Give me dialtone)
Supervisory Signal:
Used to send signals
ON HOOK/OFF HOOK/RINGING (Sent using AC current rather than DC)
Informational Signal
Dialtone/Busy/Ringback/Congestion/Reorder/Receiver Off Hook/No such number/Confirmation
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UNDERSTANDING ANALOG CONNECTIVITY
CONT..Address Signal
Dialing information over an analog line:
PULSE BREAK/CONNECTED TIP/RING
Dialtone Multi Frequency DTMF:
1209hz 1336hz 1477hz
697hz 1 2 3
770hz 4 5 6
852hz 7 8 9
941hz * 0 HASH
*DIGITS REPRESENT FREQUENCIES*
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UNDERSTANDING DIGITAL CONNECTIVITY
Problems with Analog
1. Distance limitations (Repeaters Layer 1)
2.
Wiring limitations (Messy)
*TIP AND RING FOR PAIRING KEY CONCEPT*
3. Digital voice eliminates distance issues
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DIGITIZING VOICE
Step 1 Sample the signal
If you sample the signal at twice the highest frequency, you can accurately reconstruct a signal digitally.
Common Frequencies:
Human Ear - 20-20,000hz
Speech - 200-9,000hz
Nyquist Theorum 300-4,000hz
Step 2 Perform quantization on the sample
Pulse Amplitude Modulation PAM
1. Take value of amplitude/voltage (Segments )
2. Many samples are taken as low as possible in human speech range
3. PAM scale to line up samples with voltage level
Step 3 Convert to binary
Pulse Code Modulation PCM
A-LAW (OTHER PLACES) + N-LAW (USA) - A-LAW makes more sense
Takes binary to represent POSITIVE and NEGATIVE 1 0 0 1 1 0 0 1 (2-4 bits SEGMENTS, 5-8 bits INTERVALS)
N-LAW is exact opposite (Transcoding is where you convert between the 2)
Step 4 Optionally compress the samples
1. Send all
2. Just send changes
3. Build a codebook
4.
Standard voice = 64Kbps Compressed value = 8Kbps with G.729
Example: 8000x8=64000
COW x 8000, most samples will be
the same. Hence compression!
Human voice/Codebook built as
there are only so many frequencies
used.
G.729 codec used for 8Kbps voice.
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MODERN VOICE: VOIP FOUNDATIONS
Call Processing Models
Key Voice Protocols
Deployment Models
DISTRIBUTED MODEL
Phone Session RTP
Connect Message Sent
Bridge Phone and Phone for
RTP communication
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MODERN VOICE: VOIP FOUNDATIONS
CENTRALIZED - Server and Client Model Faith in redundancy!
MGCP Protocol for
Centralized model
Routers and Phones
are workhorses
Simplicity
SRST Backup/
Failover/Mini Brain
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KEY PROTOCOLS
Signaling Protocols (Setup a call)
H.323 Peer to Peer, Between VGs
MGCP Server to Client, Between VGs
SIP Long term option/victor
SCCP Cisco Proprietary
Streaming Protocols
RTP Focus Realtime Transport Protocol/
Sound of voice
RTCP Control/Stats for call
*SIP Supports proprietary extensions
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CAMPUS IPT DESIGN
Single Site
*G.711Wideband
Codec
*ITSP No
true QoS over
WAN
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MORE DESIGN..
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WHAT IF NO WAN?
SRST When WAN fails, router takes on calls via PSTN.
PSTN should always be in place for backup.
TEHO Tail End Hop Off UK call via WAN to CHINA, then tailing off to a local call. FREE
via WAN link and only then paying a local call fee.
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DISTRIBUTED MULTI CLUSTER DESIGN
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PREPARING THE INFRASTRUCTURE FOR VOIP
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3 ROLES OF A CATALYST SWITCH
To provide Inline Power (Initially Cisco only) or Power Over Ethernet (802.3EF)
Dual VLANs/Voice VLANs/Aux VLANs (Same thing..)
Class Of Service CoS Layer 2 Markings How switch queues traffic.. + Quality
Of Service QoS Layer 3 Markings Prioritize traffic..
8 wires in standard network cable
4 used for Data transmission
PoE uses opposite 4 cables
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POWER
3 ways to power an IP Phone
Inline Power Cisco Pre Standard/IEEE 802.3AF
Midspan Power Power Patch Panel (Cost wise might as well get PoE)
Wall Power (Power supply/pack)
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POE CONFIGURATION AND COMMANDS
Show power inline BIG ONE
Switch is specd out to power every port with a phone
CDP communicates to the switch exactly how much power the phone is consuming.
Configuration
Conf t -> interface ______ -> power inline ->>> auto/delay/never
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NORMAL SWITCHING WORLD
One collision domain per port
Broadcasts sent to all ports
One subnet per lan
Limited access control
Vlans logically group users
Segments broadcast domains
Subnet correlation
Access control
QoS
VLANs traverse switches via trunks
Switch adds tag with VLAN id
TAG is removed before hitting PC
Only across trunks (To assist QoS)
Flexability (VLANS)
Segmentation of users without routers (Layer 2)
No longer limited to physical location
Tighter control of broadcasts
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VOICE/AUX VLANS
General network design/security dictates voice and
data separation.
Seems impossible since IP phones have a built in
switch.
VOICE VLANs always LOW VLAN ID As STP will
failover the lowest VLANs 1
st
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PREPARING THE INFRASTRUCTURE FOR VOIP
PART 2IP Phone Boot Process
1. Cisco switch detects PoE capabilities. (Inline or 802.3af)
2. Switch sends voice VLAN via CDP to phone.
3. IP Phone sends DHCP discover and receives a DHCP offer including option 150 (IP
address of TFTP server)
4.
IP Phone contacts TFTP server and receives configuration file.
5.
IP Phone registers with CME router.
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DHCP SERVICES ON A ROUTER
1. Excluded any necessary IP addresses (1-10 is best practice and/or 245-254)
2.
Create DHCP pool
3. Define network
4. Define Default Router
5. Define DNS
6.
Define any other options (150)
7. Configure IP helper addresses if needed
*Option 66 = Option for TFTP by name rather than IP address
*IP Helper required for phones to obtain DHCP via highest Layer 3 interface
*show ip dhcp binding
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NTP SERVERS
1. Configure NTP server
2.
Optionally confogure one of more of your devices as NTP masters.
Ntp server _ _ _ _
Clock timezone NAME hoirs offset from UTC
*show ntp associations
Designate CUCM as NTP master
Set from Stratom 1 Server
Ntp master
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CISCO CALL MANAGER EXPRESS GETTING
FAMILIAR WITH ADMINISTRATION CME Administration options
CME command line
CCP
ADMIN - OS CLI (Jeremy Cioara preferred)
CME GUI from router flash
CCP
Conf t
Telephony-service ? LOTS OF OPTIONS
Show ephone registered
Show ephone section ephone-dn
SKY IS LIMIT
CLI Dial Peers/Phonebook/Route Plan
CME GUI Phased out Router GUI = RUBBISH Evolve to CCP
CCP = Nextgen for SDM
Features Wizard to setup router as CUCME
Telephony settings have to be setup
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CME EPHONES AND EPHONE-DN
Ephones-DNs are representations of directory numbers.
Can be single line or dual line (two simultaneous calls)
Configuration
CME Router
Show ephone MAC/SIGNALING PROTOCOL/ID
Show run include ephone
Conf t
Ephone-dn (1-150) tag single line/dual line
*You cant flip modes, you have to delete and reapply
Router provisions resources
Ephone-dn 1 number 1001
Secondary Numbers
Ephone-dn 1 number 1001 secondary 10001001
Ephones
Representation of Cisco IP Phone
Linked to device by MAC ADDRESS
1. Printed on box of phone
2. Printed on back of phone
3. Settings>Network Configuration Menu
Show ephone
*Auto registration by default
Conf t
Ephone 1
Mac-address ____ ____ ____ ____
Type _____ (OPTIONAL)
*Add phone by MAC so router doesnt forget
Configure ephone + ephone DNs
1. Configure necessary EPHONE-DN
2. Configure necessary EPHONE
3.
Associate EPHONE + EPHONE-DN using the
BUTTON COMMAND (Next slide..)
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BUTTON COMMAND - BASICS
CIPC = Soft Phone (Cisco IP Communicator)
Under ephone:
Button ?
Button 1:2
1=BUTTON 2=EPHONE-DN 2
Then restart or reset.. (Restart = WARM BOOT, Reset = HARD RESTART)
Button 2:2
Button 3:3 etc
Button 1:1 2:2 3:3 One line of configuration.
At this point the phone is working on the network
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MORE BUTTON COMMAND MADNESS!
CISCO CME CISCO CONFIGURATION
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CISCO CME CISCO CONFIGURATION
PROFESSIONALExam heavy
**Service contract required with smartnet agreement**
Configuration document on Cisco website (Also prerequisites)
HTTP Based/Local Authentication user account
2 flavors of CCP:
Express running from the router flash
CCP Full Suite on PC
Discover device before use
Unfied Communications
Telephony settings to be setup 1
st
CISCO CME CISCO CONFIGURATION
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CISCO CME CISCO CONFIGURATION
PROFESSIONALSteps via CCP
1. DNs and Phones (Any Order)
2. Links with user account (User unites DN + Phone)
Add Phone
Type
MAC
Autoline (Active Line)
Add DN
Primary Number - DN
Secondary Number DDI
Name
Description
*E.164 Registration Register with SIP Provider - ITSP
Add User
User ID
Name
Display Name
Pwd Generation: CUSTOM
PIN Generation: Blank
LINK PHONE+DN TO USER
CISCO CONFIGURATION PROFESSIONAL
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CISCO CONFIGURATION PROFESSIONAL
PART 2 - FEATURESCME Features
Phone Directory
Forwarding
Transfer
Call Park
Call Pickup
Intercom
Paging
After Hours Restrictions
Single Number Reach
CISCO CONFIGURATION PROFESSIONAL
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CISCO CONFIGURATION PROFESSIONAL
PART 2 - FEATURES
Phone Directory
Button press on phone
-Personal Directory (Personal to user)
-Corporate Directory
-When an extension is created in CCP it is auto populated
into the Corporate Directory.
-Advanced -> Directory Naming Schema
-Telephony Settings/Directory Services Add manually
and limited to 100
CISCO CONFIGURATION PROFESSIONAL
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CISCO CONFIGURATION PROFESSIONAL
PART 2 - FEATURES
Forwarding
Extension -> Advanced -> Call Forwarding
Toll Fraud Call forward/Max length (Stop international
calls on call forward)
Transfer
Transfer pattern
CCP Advanced Telephony
*Transfer to non Cisco phones*
9 _ _ _ _ _ _ _ _ _ _ (10 digits allowed only)
CISCO CONFIGURATION PROFESSIONAL
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CISCO CONFIGURATION PROFESSIONAL
PART 2 - FEATURES
Call Park
-Cool
-Park call at phone number rather than extension.
-Telephony Features -> Call Park -> Create -> Name
-Number of slots for parking, reminders etc Lots of
advanced features
CISCO CONFIGURATION PROFESSIONAL
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CISCO CONFIGURATION PROFESSIONAL
PART 2 - FEATURESCall Pickup
Group of people in same team. Call can be picked up by any phone/
DN.
Pickup group can be any number
Searches group only
Telephony features->Call Pickup Groups->Create->Add Extensions,
also Softkey on phone.
*CUCM provides lots more configuration for pickup groups.
Intercom
Bridge/button setup for intercom call. A directly to B.
Used with directors/CEO etc
Whisper Mode Intercom Auto Answer on speaker phone
Can be set as a speeddial/Label Button
Dedicated 2 way audio path between 2 phones
CISCO CONFIGURATION PROFESSIONAL
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CISCO CONFIGURATION PROFESSIONAL
PART 2 - FEATURES
Paging
Make an announcement using phone system to all phones
All phones will go into speaker phone mode
CUCM never really had this feature, but CME has it
Paging Numbers Name/Description/Number/Members
(239.0.0.1:2000 UDP)
Paging Groups Groups of Groups
After Hours Restrictions
Not exam important.. Bonus
Allow or deny certain numbers during certain times.
Telephony features->After Hours Toolbar
Prefix Block/Schedules etc..
CISCO CONFIGURATION PROFESSIONAL
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CISCO CONFIGURATION PROFESSIONAL
PART 2 - FEATURES
Single Number Reach
People can reach you by dialing 1 number only.
Extensions->Advanced->Single Number Reach
Remote Number/Time (Seconds)/Timeout Value
GATEWAYS AND TRUNKS: UNDERSTANDING
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GATEWAYS AND TRUNKS: UNDERSTANDING
VOICE CODECSDigital conversion process
*Nyquist Theorum* Analog waveform/signals which are converted to binary.
How to turn spoken voice into bits with 4 steps.
Step 1 Take many samples of the analog signal
Step 2 Calculate a number representing each sample (aka QUANTIZATION
Pulse Code Modulation)
Step 3 Convert number to binary
Step 4 (Optional) Compress signal
GATEWAYS AND TRUNKS: UNDERSTANDING
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GATEWAYS AND TRUNKS: UNDERSTANDING
VOICE CODECSCommon Audio Codecs
G.711 64Kbps MOS = 4.1
G.729 8Kbps MOS = 3.92 **NO 1 CODEC**
G.729A 8Kbps MOS = 3.7 **NO 2 CODEC**
G.726 32Kbps MOS = 3.85
G.728 16Kbps MOS = 3.61
ILBC Internet Low Bitrate Codec 15.2 Kbps MOS = 4.1 NEXT GEN + OPEN
SOURCE
MOS = Mean Opinion Score - BAD 1 5 GOOD
Normal PSTN = MOS 4.0 4.1
*Each channel/DSO consumes BW value with all headers adds to 80Kbps.
GATEWAYS AND TRUNKS: UNDERSTANDING
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GATEWAYS AND TRUNKS: UNDERSTANDING
VOICE CODECSChoosing a codec and sample size
Sample size dictates the amount of audio included in each packet. (Default = 20MS
of audio)
Larger samples = bandwidth samples
Larger samples = more delay
Bytes per sample = (Sample size * Codec Bandwidth) / 8
GATEWAYS AND TRUNKS: UNDERSTANDING
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GATEWAYS AND TRUNKS: UNDERSTANDING
VOICE CODECSAdding in data link/network overhead
Ethernet = 18 bytes
Frame Relay = 4-6 bytes
PPP/MLPPP = 6 bytes
-
IP = 20 bytes
UDP = 8 bytes
RTP = 12 bytes N + T = 40 bytes
Tunneling Bonus Overhead
GRE/L2TP = 24 bytes
MPLS = 4 bytes
IPSEC = 50-57 bytes
Adding it all together
Total Bandwidth = Packet Size*Packet Per Second
Packet Size = 218 bytes
Packets Per Second = 50 x 218 = 10900 bytes per sec
10900 x 8 = 87200 bps/1000 = 87.2 Kbps of BW ---- WOOOOOOAH
GATEWAYS AND TRUNKS: UNDERSTANDING
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GATEWAYS AND TRUNKS: UNDERSTANDING
VOICE CODECS
VOIP BANDWIDTH SAVINGS MEASURES
1. Voice Activity Detection (VAD): Suppresses the silence in the
conversation. Average of 35% BW savings.
2. Compressed RTP: Compresses network and transport layer
headers from 40 bytes to 2-4 bytes.
Bandwidth savings are codec dependent. (Around 40% with G.729
CODEC)
Option 2 is processor intensive
GATEWAYS AND TRUNKS: UNDERSTANDING
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DIGITAL SIGNAL PROCESSOR RESOURCES
Digital Signal Processors Offload media processing function from
voice processing equipment to dedicated hardware chips.
- Coding
-
Transcoding (One CODEC to another)
- Media Termination Point (MTP)
-
Conferencing Router = MIXER PSTN+VOIP into 1 stream.
VOICE TO PACKETS
2 forms of DSP = C549 and C5510
CODEC COMPLEXITY
G.711 - MEDIUM
G.726 - MEDIUM
G.729A - MEDIUM
G729AB MEDIUM
G.723 HIGH
G.728 HIGH
G.729 HIGH
G.729B HIGH
ILBC HIGH (DSP Calculator on Cisco Website)
GATEWAYS AND TRUNKS: UNDERSTANDING
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GATEWAYS AND TRUNKS: UNDERSTANDING
DIGITAL SIGNAL PROCESSOR RESOURCESRTP and RTCP
RTP is a QoS consideration
RTP carries audio payload between devices
RTCP carries call statistics between devices
RTP uses random, even numbered UDP ports between 16384-32767
RTCP uses random, odd numbered UDP ports between 16384-32767
GATEWAYS AND TRUNKS: CONNECTING CME TO
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OTHER VOICE SYSTEMS
CME to LAN
CME to PSTN
CME to PBX
CME to PSTN VOIP
Voice Gateway Types
Analog voice gateway One call per port
Digital voice gateway Multiple calls per port
A voice gateway transitions between voice network types (VOIP/PSTN)
Same concept as a router separating networks.
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GATEWAYS AND TRUNKS: FXO AND FXS
FXO+FXS
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GATEWAYS AND TRUNKS: DIGITAL VOICE PORTS
Voice or data = VWIC 2MFT T1/E1
Card is beefy
24 channels T1
32 channels E1
T1 and E1 Common Associated Signaling (CAS) Most common
RBS
T1 and E1 Common Channel Signaling (CCS) Primary Rate
Interface (PRI)
Basic Rate Interface (BRI) = 2 channels of voice, 1 for signaling
CCS provides a dedicated channel for signaling.
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GATEWAYS AND TRUNKS: VOICE GATEWAYS
Gateways change
between VOICE
and DATA.
Gateways bridge
communications.
Gateway Control/
Signaling
Protocols
H.323 (Default/
Old) Audio/Video
Comms Suite
MGCP Used
primarily by Cisco
with Server +
Client model
SIP Poised to be
the universal VOIP
standard
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GATEWAYS AND TRUNKS: SIP
Designed as next generation H.323
Call Signaling and Call Setup
Avaya use SIP all the time
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CME DIAL PEERS: PART 1
Types of dial peers:
POTS Dial Peers
Connect to any traditional telephony network or devices
Defines number reachable through a given PORT (Keyword)
VOIP Dial Peers
Connect across any packet based network
Defines number(s) reachable at a given IP address
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CME DIAL PEERS: PART 1
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CME DIAL PEERS
CME DIAL PEERS
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POTS + VOIPExample
CME DIAL PEERS
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CME DIAL PEERS
Show voice port summary
Ports->Sig Type->In Status->On/Off Hook
Conf t
Dial-peer voice tag type VOIP/POTS
Destination pattern 3301
Port 1/0/0
Dial-peer voice tag type VOIP/POTS
Destination pattern 3302
Port 1/0/1
*This enables the 2 POTS phones to communicate through the CME router/FXS ports.
Useful commands:
Debug voip dialpeer
Show voice call summary
CME DIAL PEERS VOIP
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CME DIAL PEERS: VOIP
Conf t..
Dial-peer voice 330 voip
Destination pattern 330.
Session target ipv4: 10.1.1.2
*Default codec used is G.729*
1
st
Call Leg over IP to Voice Gateway
Show dial-peer voice summary
CME creates dial peers for all registered phones.
A target is required:
Session target ipv4: 10.1.1.2
VAD = Voice Activity Detection
Show dial-peer voice summary
CME DIAL PEERS WILDCARDS
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CME DIAL PEERS: WILDCARDS
Period (.) = 1 digit
Plus (+) one or more proceeding digits
Brackets [ ] = Range of digits
Example [1-3] 1111, 2111, 3111
T = Any number of digits (0-32) *Generic wildcard
DP-9T (Anything up to 32)
Lazy dial plan 9.anything up to 32.
Dial-peer voice 10 voip
Destination-pattern 10.. (10XX)
Session target ipv4: 10.1.1.1
Show dial-peer voice summary
1005 . Match 1005 dial peet tag id
*IMPORTANT NUGGET*
CME DIAL PEERS PART 2
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CME DIAL PEERS: PART 2
PSTN wildcards are out of CCNA Voice scope.. Phew
Dial peers TO the PSTN are included
CAS = Stealing bits/CCS = Dedicated
LEGACY VOICE ROUTER
Show voice ports summary
T1 = WIC Data or Voice
You need to tell the router which..
CONFIGURATION
Controller T1 1/0
Framing esf (USA)
Linecode b825
Ds0-group 5 timeslots 1-24 type fxo GROUND or LOOP *NOT IN CCNA
CME DIAL PEERS PART 2
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CME DIAL PEERS: PART 2
Show voice port summary
*Will display 24 FXO loop start ports
Pri-group timeslots DIGITAL PORT CFG
CCNA Voice you are only expected to know DIAL PEERS FOR PSTN
Destination-pattern 9T (PSTN WILDCARD DIAL PEER)
*Never know when done
*destination-pattern [2-9] (7 digit)
Wildcard for area code/local prefix destination-pattern [2-9]..
Dial-peer voice 9 POTS
Destination-pattern 9[2-9].. [2-9]
Port 1/0:5
Dial-peer voice 91 PORTS
Destination-pattern 91 etc (USA BIASED, NEED UK EXAMPLES)
CME DIAL PEERS OUTBOUND DIAL PEERS
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CME DIAL PEERS: OUTBOUND DIAL PEERS
**The 555[1-3]
Session target ipv4: 10.1.1.1
Dial-peer voice 2 voip
Destination-pattern 5551
Session target ipv4: 10.1.1.2
Dial-peer voice 3 voip
Destination-pattern 5551
Session target ipv4: 10.1.1.3 --------This dial peer wins.
*Add a T . For 0-32 number of digits, also a # to process the call
immediately.
CME DIAL PEERS: INBOUND DIAL PEER
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MATCHINGNext call leg has to have a dial-peer to know what to do. (IN and OUT)
MANIPULATING DIALED DIGITS
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MANIPULATING DIALED DIGITS
Auto stripping rule of POTS dial peers
POTS dial peers automatically strip any explicit defined number
from the destination pattern before sending the call.
Any non wildcard number specific only (EXPLICIT)
DIGIT MANIPULATION COMMANDS
1. Prefix (Add digits to left)
2. Forward-digits (How many digits?)
3. Digit-strip Turn off stripping
4. Num-exp (Match X and change to Y)
PSTN FAILOVER
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PSTN FAILOVER
PSTN FAILOVER
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PSTN FAILOVER
Dial-peer voice 6000 VOIP
Destination-pattern 6
Session target ipv4: 10.1.1.2
*preference 0
Dial-peer voice 6001 POTS ----- PSTN BACKUP
Destination-pattern 6 (STRIPPED)
Port 1/0:1
No digit-strip
Prefix 1512555 (USA LONG DISTANCE)
PSTN DIRECTING CALLS TO RECEPTIONIST
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PSTN DIRECTING CALLS TO RECEPTIONIST
PSTN EMERGENCY CALLS
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PSTN EMERGENCY CALLS
CLASS OF RESTRICTION
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CLASS OF RESTRICTION
Calling privileges
ACLS for VOIP
CCNA Voice COR concepts only not configuration
Who can call what?
PBX Realm Class of Service (NOT QOS)
Call manager realm Class of Control
Router realm = Class of Restriction
Requires an in depth understanding of in + out dial peers
Requires more detailed dial peers (No 9T for PSTN)
Manually creating a PSTN dial plan
EMERGENCY COR
Dial-peer voice 999 pots
Destination-pattern 999
No digit-strip
Forward digits 3 (3 far right digits)
INTERNATIONAL
Dial-peer voice 12 pots
Destination-pattern 9011T
Prefix 011 (USA BIASED)
CLASS OF RESTRICTION
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CLASS OF RESTRICTION
COR LISTS
Incoming dial-peer assigns incoming COR list
Outgoing dial-peer assigns outgoing COR list
If the OUTGOING COR list is a subset of the INCOMING COR the call IS forwarded
UNDERSTANDING COR *NOT FOR CCNA VOICE
*Bubble analogy
Dial-peer cos custom
Name 911 call
Name local call
Name ldcall
Name international
CLASS OF RESTRICTION
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CLASS OF RESTRICTION
STEPS FOR COR
1. Define COR bubbles under cos custom NAME
2. Define outgoing COR lists (MEMBERS)
3. Define incoming COR lists (MEMBERS)
4. Assign COR to dial-peers
WITHOUT INCOMING COR LIST YOU CAN DO ANYTHING
1. Lists and names
2. Outgoing bubbles
3. Incoming bubbles
4.
Apply COR to dial-peer corlist incoming LD
CCNA VOICE FOCUSES ON CONCEPTS ONLY NOT CONFIGURATION
CISCO UNIFIED COMMUNICATIONS MANAGER
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OVERVIEW
!CUCME = CM on single voice gateway
!
CUCM = Redundancy, scalable etc..
!CUCM doesnt interface to PBX, hence voice gateway
requirement between digital and analog worlds.
!Call Manager= The mind of the voice network
!Major Functions: Call Processing, Signaling and Device
Control, Dial Plan Administration, Phone FeatureAdministration, Directory Services and Link to External
Applications.
CUCM HISTORY
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CUCM HISTORY
! Version 2.4
! Cisco Made Own
! NT Based 4.0
! 2001
! Install on any hardware
! Cisco blamed for faults!
!
Version 3.0
! Only Cisco approved hardware
! Media Convergence Server
! BUT if not purchased from Cisco, NO end to to end support.
!
Version 4.x 2000
! Version 4.3 2003
! Version 5.x Linux Build/2003
! Version 6.x Cisco stood ground on Linux based OS *MAINSTREAM*
CUCM - FEATURES
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CUCM - FEATURES
!HTTP is 90% of all administration
!
IE and Firefox only, doesnt like Chrome!Navigation URL/cucmadmin (5 consoles)
!System Menu = Global Configuration Mode equivalent
!Serviceability Menu = Monitoring/Alarms/Tools/Features/Services
!
Control Center = Start/Stop Services (Features/Network)!OS Administration = Tether to OS
!Disaster Recovery System = Backup/Restore CUCM database only
!OS Administration to update version of CUCM
!Cisco Unified Reporting = Reports/Data from CM/Sucks data from
all clustered CMs.
!All services are installed by default, just activate and deactivate as
required.
CUCM - CLI
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CUCM CLI
!You can SSH into the CUCM server
!LAB CUCM with VMWare and CUCM ISO
!SSH = Overlay of Linux OS (Restricted)
!
Utilities PING example
!Database Replication
CUCM SUPPORTING END DEVICES
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CUCM SUPPORTING END DEVICES
DEVICE POOL
!
Assign settings to phone
!Assignment to IP Phone
!
List of CUCM servers to use!Codec to be used
!Time + Date information
!
DEVICE POOLS group thisconfiguration to a single assignment
CUCM SUPPORTING END DEVICES
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CUCM SUPPORTING END DEVICES
REQUIRED DEVICE POOL ELEMENTS
!
Device Pool NAME!Cisco CM Group (Up to 3)
!Date/Time Group
!
Region!Softkey Template
!SRST Reference
!*DEVICE POOL is normally set as a LOCATION
!
By default the CUCM group only contains the
PUBLISHER
!All auto registration devices will go to this
PUBLISHER/CUCM GROUP
CUCM DEVICE POOL ELEMENTS
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CUCM DEVICE POOL ELEMENTS
DATE/TIME
!
CM Local = Default Greenwich Time!Create Timezone
REGION
!G.711 = Uncompressed 64Kbps per call
!
G.729 = Compressed 8Kbps (20ms delay)!G.729 offers human dictionary, MOS Scale/Score,
Already covered!
!Different regions can use different codecs, dependent
on their bandwidth capabilities.!Relationships can be setup between regions
!Phones are added to the required region via DP
membership.
CUCM DEVICE POOL ELEMENTS
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CUCM DEVICE POOL ELEMENTS
SOFTKEY TEMPLATE
!Dictates what keys are available on the IP phone!
Device Device Settings Softkey Template!You cannot change the default softkey template!You can copy a softkey template!Softkey Layout GO!Undefined Key = BLANK
SRST REFERENCE
!Disabled by default or use default voice gateway!If router runs SRST, router supports phone and can talk to
PSTN/other site.!
Voice Gateway runs SRST!System->SRST!*2000 = PORT
!Pool is ready for devices to assign to
!Pool created per location
CUCM SUPPORTING END DEVICES PART 2
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CUCM SUPPORTING END DEVICES PART 2
MANUALLY
!
Enter MAC address and Directory Number for each phone. (UPCcode scanable)
!Selsius Ethernet Phone (Selsius are an organization)
!PHONE BUTTON TEMPLATE: controls line buttons
!DEVICE SECURITY PROFILE: Encryption settings
!
Add live buttons: DN 2001!*HTTP Access to phone via HTTP server SECURITY CONCERN!!
AUTO REGISTRATION
!CUCM hands out extensions to newly registered phones, similar to
DHCP.!Default configuration file
!Static Assignment
!Good for new delivery
CUCM BULK ADMINISTRATION TOOL
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CUCM BULK ADMINISTRATION TOOL
BULK ADMINISTRATION TOOL
!
Use an Excel spreadsheet to generate CSV file of devices.!System->Unified CM->Enterprise Parameters->SCCP for auto
registration
!Device->Device Defaults>All Defaults for auto registration
BAT
!
Phones->Phone Template (Generic) Save Template (Sales Example)!Add Line Template also Directory Number specified
!Spreadsheets Macros Enabled
LOCKING DOWN THE CISCO IP PHONE
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LOCKING DOWN THE CISCO IP PHONE
!Disable PC port
!
Lock settings access!Gratuitous ARP protect
!PC Voice VLAN access
!
IP Phone HTTP access!Product specific configuration layout
!GARP = ARP comes in that you didnt ask for.
GARP sends a fake MAC address for your default
gateway. Disabled in CUCM7.!Phone conversation sent to PC and SWITCH.
Call recording/monitoring etc.
CUCM SUPPORTING END USERS
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CUCM SUPPORTING END USERS
BENEFITS: Users can manage phone/Soft phones requires logins
! Advanced Features: Extension Mobility
! Tracking Per User Account
!
There are 2 different types of users, END USERS and APPLICATION USERS
! End users can be linked to LDAP (Optional)
CUCM can use 3 LDAP Options:1.Local data only (NO LDAP)
2.LDAP Sync
3.LDAP Authentication
LDAP Sync
! Disables bulk of CUCM User Management (Read Only)
! Passwords/CUCM specifics managed from CUCMLDAP Authentication
! Passwords managed in LDAP not in CUCM
! Authenticates directly against LDAP DATABASE
! LDAP SUPPORT MS/NETSCAPE/SUN/IPLANET
! Must setup a SYNC AGREEMENT between CUCM and LDAP
! User Search Base and User ID LDAP attribute
! LOCAL LDAP User Management->App User->End User Add New Fields
! Associate END USER with device
! Now the user can login and manage associated devices.
CUCM BULK ADMINISTRATION TOOL PART 2
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!Used for large additions or changes to CUCM Database
!
Phones/Users/Many tedious configurations!Pre-integrated in CUCM Administration
!Export and reimport
!Exported data can be used for inplace migration or data
restore (Not possible with DRS)BAT COMPONENTS
!Template (Phone/Users) and CSV file -> BAT Engine
!Bulk Admin Excel Template/Upload or Download
Files/Download and open in Excel/Enable Macro/RunMacro and export to txt (CSV)
!Create User Template Name Sales (Example)
CUCM MORE LDAP
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LDAP
! Supported Directory Active Directory
!
Setup SYNC AGREEMENT! Create Service/User Account in AD
! Create OU for sync
! Serviceability > Service Activation-> Enable Cisco DIR SYNC
! System->LDAP->LDAP System
!
Enable Sync with Active Directory
!
LDAP Attribute SamAccountName/Email etc..! LDAP = READY
! Add new LDAP directories:
! LDAP Manager = username@domain
! LDAP User Search Base - = OU (LDAPSYNTAX=OU=CCMEndUsers,dc=home,dc=local)
!
Add more than 1 DC.! LDAP AGREEMENT in place, perform full sync.
! LDAP Admin done from AD.
! Change passwords from CUCM or AD.
! LDAP Authentication = Same setup as LDAP Sync.
! One in place, all password changes done from Active Directory.
CUCM: MANAGING GROUPS, ROLES AND
PRIVILEGES
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PRIVILEGES
!Delegate administrator rights
!
Users assigned to groups
!Groups assigned to one or more roles
!Roles assigned to privileges
!
*Important ordering!
CUCM UNDERSTANDING DIAL PLANS: CUCMROUTE ARCHITECTURE
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CUCM UNDERSTANDING DIAL PLANS
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!
CUCM only knows about what is in the database
!
No dialpeer required (CME configuration)
!
CUCM only knows whats in cluster *1 PUBLISHER per cluster*
! ROUTE PLANS = Required for out of cluster communications.
CUCM DIAL PLANS CONFIGURATION
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!Add DEVICES VG and PSTN
!Call Routing -> Route Hunt -> GROUP/LIST/PATTERN
!Distribution Algorithm Circular or From Top (PREFERRED)
!Add Route List Ways to 2xxxx Groups to be used for calls
!When WAN call sent over PSTN requires TRANSFORMING *ROUTE
LIST LEVEL*
!
PATTERN *LINK TO GATEWAY/ROUTE LIST!Ties everything together
!2xxx WILDCARD
!X = SINGLE DIGIT
!@ = North American Numbering Plan
!
! = One or more digits (32 digit cap)
! . = Access Code Termination
!HASH = Terminates Interdigit Timeout
!Provide outside dialtone
! Predot Strips before .
CUCM WILDCARD SAMPLES
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![XYZ] [X-Y] [X-YZ] DIGIT SET
!
[XYZ] [X-Y] [X-YZ] Negative Digit Set! = Anything BUT!
!
Example of each:!38[2,4-6,9]3 = 3823
!38[2-4]3 = 3813
!9011!HASH =
9011_______________________HAS (32 digitsthen terminate)
CUCM PARTITIONS AND CALLING SEARCH
SPACES
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SPACES!
Restricting devices from calling certain numbers
PARTITIONS
! Groups of dialable numbers - Lines/Route Patterns/Anything that has a number
! Examples: LOCAL-PT, INTERNAL-PT, INT-LD-PT
CALLING SEARCH SPACES
!
A list of reachable partitions
!
Assigned to any dialing entity
! Defines calling privileges
! Examples: INT-CSS contains INT-PT, LOCAL-PT partitions
! Phone in the internal partition doesnt dictate calling privileges, this is where the
CSS kicks in.
!
Partitions and CSS SIDE BY SIDE (GROUP and PRIVILEDGES)
! By default all phones+numbers are assigned to the NONE partition and CSS.
!
Everything can call everything by default.
!
Directory Number is in the partition not the phone.
! Bottom of all CSS is NONE partition. Best practice is to leave nothing in the NONE
partition or CSS.
CUCM PARTITIONS AND CALLING SEARCH
SPACES
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SPACESExample
!
3 types of calling restrictions should exist in your organization:
1.
Lobby/public phones: Internal Extensions only
2.
Typical Users: Internal and Local PSTN
3.
Management: Internal, Local PSTN and Long Distance PSTN
!
STEP 1 Create the partitions
!
STEP 2 Assign numbers to partitions
! STEP 3 Create Calling Search Spaces
! STEP 4 Assign Calling Search Spaces to Devices
CUCM PARTITIONS AND CALLING SEARCH
SPACES
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SPACES
CUCM FEATURE OVERVIEW
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! Phone feature madness!
! Cisco IP Phone media streaming application IMPORTANT?
CALL PARK! Call Routing-> Call Park -> Add a range of numbers: 115x (0-9 Slots to Park call)
! Service Parameters -> Long List/Overview of a lot of odd parameters! Call Park in the list/Settings for Call Park.
CALL PICKUP
! Call Routing-> Call Pickup Group-> Add New Name + Number
! Group Call Pickup 0 Pickup
SHARED PHONE LINES! Add DN to 2 registered devices
! Multiple call waiting settings under DN
! Number of calls per device Max number can be set but 196 MAX
! Busy Trigger 2 people online, 3rdwill be busy.
! Any change to DN impacts all devices. EDIT LINE APPEARANCE.
DO NOT DISTURB
!
Modify softkey template to enable DND feature
! Copy template to DND user
! Configure softkey layout
! On Hook
! Toggle DND
! Phone Configuration Menu On and Off Tickbox
CUCM FEATURE OVERVIEW CONT
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CALLBACK
! Lift handset and hungup
!
User notifed that a user is available.! Add Softkey under On hook
! Goes into effect on ring out of calling phone
! Crafty! Plays a chime when user is available.
BARGE AND PRIVACY
! User can join a call (BARGE)
! Shared Lines
!
Privacy button can prevent BARGING (DEFAULT)! Device->Phone->Per Phone or Cluster
! Built in Bridge = BARGE
! On the fly, bridged conference call
! Phones handle conference call themselves
! Service Parameter Configuration = Global Settings for BARGE.
SERVICES/EXTENSION MOBILITY
! Custom programs for phone.
! IP Phone Services Configuration
! Point to URL for apps
! Subscribe to this service under Device->Phone
! EXTENSION MOBILITY is covered in CCNP VOICE
! This is an XML service enabling users to login to the phone.
CISCO UNITY CONNECTION
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FOCUS
!
One of 5 LINUX VOIP appliances
!
Integrates with legacy PBX via PIMG or TIMG
! USERS Manual, CSV, CUCM Import or LDAP
! CUC integrates with CUCM using SCCP or SIP
! PIMG Up to 8 digital or analog ports
! TIMG Digital T1 to SIP
!
SCCP = Easier to setup than SIP (Jeremy opinion!)
CISCO UNITY CONNECTION CONT..
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!
SIP Trunk used for CUCM and CUC Connection (Destination Address)
!
SCCP requires message waiting (Integrated with SIP)
!
CUCM Admin -> Voicemail! Wizard based
HOW CUC PROCESS CALLS:
! Call Handlers Scripting language for Unity
!
System Greetings/IVR Equivalent/Series of Handlers
!
Directory Type Users DN to reach them
! Interview Literally an interview, answer questions etc.. Info collector resource.
! Calls are identified as direct or forwarded
! DIRECT CALLS Messages button Calls Unity to collect VM
! FORWARDED CALLS CFNA, CFB, DND, Auto Attendant Forwarded when N/A
CISCO UNITY CONNECTION CONT..
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! Managing user and mailboxes in CUC
! User templates, make life easier!
!
Lots of options
BASIC ELEMENTS
! -User
! -Phone: Dialing Restrictions, CoS, Schedule
!
-Location: Geographic location, language, time zone
! -CoS defines many options (Timers, Features, Restrictions)
! -You can create end users from the template+associate the phone
extension number.
! -Phone extension used to identify the user when they press the Messages
button! -User template basics Core settings
! -If no VM box you can go to default greeting,
! Import Users->LDAP->Active Directory DV or AXL Remote (CUCM TO CUC)
! Service need activating for AXL (Serviceabilty Service Activation)
!
LDAP and AXL
CISCO UNIFIED PRESENCE SERVER
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CISCO UNIFIED PRESENCE SERVER
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CISCO UNIFIED PRESENCE SERVER
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