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Thursday, January 22, 2015

CCNA Voice: Managing Users and Devices with CME

Managing Users and Devices with CME

Three key items needed to get CME configured:

  • IP Source Address
  • Max-DN
  • Max-Ephones

IP source address determines what interface will expect IP phone registration requests

Max-DN / Max-Ephone configuration reserves resources on the router and max-ephone should not exceed number of licenses purchased

Ephone / Ephone-DN Configuration

Can be configured as single-line, dual-line or octo-line

single-line ephone-dn: Can only make or receive a single call at a time. If in use caller will receive a busy signal

dual-line ephone-dn: Phone can handle two simultaneous calls and supports features such as call waiting, conference calling and warm transfer

octo-line ephone-dn: Typically use for shared lines where many share the same extension or receptionist phones

Configuration Example with CLI


config t
ephone-dn 1
number 2000
ehone-dn 2 dual-line
number 2001 secondary 2085551212


ephone 1
mac address  00ab.5454.65ba


ephone 1
button 1:2

button command links ephone-dn 2 to button 1 on ephone 1

restart command tells phone to do warm reboot and redownload configuration file from TFTP server

Button assignments can be verified with show ephone command

Configuring Users, Phones and Extensions with CCP

Telephony Service is configured from the Configure > Unified Communications > Telephony Settings

Same three key items are required:

  • Max-Ephones
  • Max-DNs
  • IP Source Address

Once telephony services are activated, CCP can be used to configure users, extensions and phones

Configure Extensions

Configure > Unified Communications > Users, Phones and Extensions > Extensions

Configure DN, description, secondary DN if applicable, line type, click OK

Configure Phones

Configure > Unified Communications > Users, Phones and Extensions > Phones

Configure model of IP Phone, MAC address, click OK

Configure Users

Users must be created to link DN to IP Phone together with CCP

Configure > Unified Communications > Users, Phones and Extensions > User Settings

Add details:

  • User ID (Only field required)

Other optional fields:
  • First and last name
  • Display Name for Caller ID
  • Password
  • PIN

Click Phones/Extensions tab to associate user with phone and extension(s) via drop-down boxes

Click OK

CCP has template of show commands via drop-down box for troubleshooting, also functions as free-form typing box for show commands that can be accessed via:

Configure > View > IOS Show Commands

CCNA Voice: Introduction to CME Administration

CME Administration

Command Line Management

Use one of three methods to access CLI:

  • Console Port
  • Telnet
  • SSH

telephony-service configuration command activates CME functionality on a router that supports it

Core CME config commands are performed under telephony-service configuration

show ephone registered command shows phones registered with CME and is most common verification/troubleshooting command

GUI Management

Two flavors: Integrated CME GUI and Cisco Configuration Professional

Integrated CME GUI

Files loaded into flash of router, with assigned IP and http server service turned on the router

Focused on telephony, not pretty


Can configure all major elements of routers


Local install on client PC, can be used to manage any supported Cisco platform

CCP Configuration

Before managing devices a community must be configured

Community consists of devices to be managed

Devices to be managed must be configured with four things to support CCP control:

  1. Reachable IP from CCP
  2. Level 15 Username/password on device
  3. HTTP services turned on
  4. Local authentication
CCP uses Telnet/HTTP by default, can be configured to use SSH/HTTPS

When a device is discovered by CCP it populates CCP with info

Unified Communications can be configured on a router (if not already enabled) in one of four ways:

  1. CME Standalone
  2. Voice Gateway (PSTN to VOIP or analog to digital)
  3. CME as SRST (CME acts as failover device if CUCM communication is lost)
  4. None
CCP has a configuration confirmation screen that shows what commands are to be delivered to the router before being applied, in order to verify correctness prior to submitting

Saturday, January 17, 2015

CCNA Voice: IP Phone Concepts and Phone Registration

IP Phone Concepts and IP Phone Registration

Connecting/Powering Cisco IP Phones

Three sources of power for IP phones available:

  • Catalst Switch PoE (pre-standard 802.3af)
  • Power Patch Panel PoE (pre-standard 802.3af)
  • Cisco IP Phone Power Brick (facility power)

Catalyst Switch PoE

Cisco Inline Power existed before official standard 802.3af was developed and used unused pairs in ethernet cable to deliver power

802.3at power standard created to increase maximum wattage fro 15.4W to 25.5W

Power Patch Panel

Patch panels are powered and inject power onto ethernet line as an intermediary

Lower cost than switch upgrades, but switches in use must otherwise support QoS and voice vlans or the switches need to be upgraded anyway, eliminating lower cost

Inline PoE Injector is even lower cost but requires dedicated power plug for each injector, not scalable

Cisco IP Phone Power Brick

Must be purchased separately from Cisco, one per phone, requires dedicated power plug for each phone brick

If IP phones have added modules (ie sidecar) then switch PoE is no longer sufficient and a brick is needed

Voice VLAN Concepts/Configuration

Cisco IP Phones support VLAN tagging and use CDP to discover voice vlan

PC can not understand tagged frames, so IP phone must strip tags before delivery to attached PC

Phone tags its own packets with voice vlan

VLAN Configuration

  1.  Add voice vlan to switch
  2. Configure IP phone switchport with mode access, access and voice vlan numbers
  3. Enable port for spanning-tree portfast to allow IP phone to boot quickly

Cisco IP Phone Boot Process

  1. IP phone receives power from the switch or one of several aforementioned power solutions
  2. Switch delivers voice vlan info to phone using CDP
  3. IP Phone sends DHCP request on voice vlan
  4. DHCP server responds with IP address offer
  5. IP phone receives DHCP option 150 with IP address and other normal info such as gateway and DNS
  6. Option 150 directs IP phone to TFTP server address to pull configuration of the IP phone
  7. Configuration includes call processing server IPs (CUCM or CME)
  8. IP Phone attempts to register with a call processing server in order of the list in the configuration

Config files are named by phone, ie, SEP(MAC ADDRESS OF IP PHONE).cnf.xml

If this file does not exist on TFTP server, IP Phone requests XMLDefault.cnf.xml that has base configuration for auto-registration with CME/CUCM

 Configuring a Router as the TFTP Server

  1. Create DHCP scope on router
  2. Add network, default gateway, dns server (optional) and option 150 address pointing at the router's voice vlan IP
  3. Ensure the configuration files are accessible on the router for the IP Phones to download 

NTP for Cisco Devices

Accurate clocks on devices are needed for the following reasons:

  • Correct date/time displayed for users
  • Correct date/time assigned to voicemail tags
  • Accurate CDR records
  • Many security features rely on accurate time
  • Logs on routers/switches are accurate with correct time

clock set command can manually set time

Stratum of NTP server determines how far away the device is from a radio/atomic clock

ntp server (IP ADDRESS) configures the device to use a server for NTP

clock timezone (name) (UTC Offset) command configures time zone

To configure the device as an NTP server, command ntp master (stratum number) is used

IP Phone Registration

Required steps before registering:

  1. IP Phone has received power
  2. IP Phone has voice vlan information via CDP
  3. IP Phone has DHCP address and option 150 address
  4. IP Phone has downloaded its configuration from TFTP server
IP Phone configuration will list up to three call processing servers (CME/CUCM), IP Phone will attempt to register in order until it successfully registers with one

Registration is done with either SCCP or SIP depending on phone firmware

SCCP is Cisco proprietary, SIP is industry standard

Registration process is as follows;
  1. IP Phone contacts call processing server, identifies itself by its MAC to the server
  2. Server consults database and sends operating configuraton to the IP Phone including Directory Numbers, ring tones, softkey template, etc using SCCP or SIP
  3. SCCP/SIP used to use phone from that point, when IP phone buttons are pressed, handset is lifted off-hook, etc

Thursday, January 15, 2015

CCNA Voice: Unified Communications at a Glance

Unified Communications Pieces

Unified Communications Products

Core products:
  • Cisco Unified Communications Manager Express
  • Cisco Unified Communications Manager
  • Cisco Unity Connection
  • Cisco Unified Presence
Other products include Cisco Unified Contact Center Enterprise/Express, Cisco Unified MeetingPlace, etc

Cisco Unified Communications Manager Express

CME was designed for ISR G2 Routers, ISR G1 routers with proper IOS and hardware can also support CME 8.X

Key Features of CME:

  • Call control device, handles signaling, call routing, call features
  • CLI or GUI based configuration using CCP
  • Local telephone directory
  • CTI support for application integration
  • Trunk to other VOIP systems (ie, CUCM)
  • Cisco Unity Express Module direct integration with network module
CME controls almost all actions performed with Cisco IP Phones using SCCP or SIP

As user inputs to the phone, SCCP or SIP messages are sent between CME and IP Phone to determine what is happening

After call setup, RTP stream is created between two endpoints and CME is no longer involved

For calls to the PSTN, CME acts as the voice gateway and transcodes analog to digital signal using DSP/PVDM modules. During the call CME transcodes between PSTN and IP phone and can not be removed from call flow

Cisco Unity Express

Integrated hardware module for CME router to provide voicemail services. Either comes as ISM or SM. ISM is internal to CME router and uses flash memory for storage. SM is external and uses a hard drive for storage. ISM/SM replace CUE AIM and NM

CUE runs its own independent Linux-based OS which is accessible from CME router after install

Key features of CUE:

  • Voicemail
  • Auto-attendant for dial-by-name, basic operator/menu capabilities
  • IVR system with basic menu tree system, more features than Auto-attendant
  • Native T.37 Fax Processing, can receive faxes and process to user's mailbox as TIFF attachment
  • SRSV sets CUE to act as backup voicemail if enterprise Cisco Unity Connection is inaccessible
  • Standards-based SIP protocol signaling between CUE and CME

Cisco Unified Communications Manager

CUCM is the call processing director of a Unified Communications solution

Key Features of CUCM:

  • Complete audio/visual telephony support
  • Appliance-based, meaning the operating system is secured/inaccessible 
  • Redundant servers
  • Intercluster/voice gateway control/communications
  • Disaster Recovery System
  • VMWare virtualization support
  • LDAP/Active Directory integration support

CUCM Database Replication

CUCM IBM Informix Database includes info such as dirctory numbers, route plan, hunt groups, etc which is replicated to all servers in cluster

CUCM Runtime (real-time) data is replicated to other cluster members using Cisco proprietary Intracluster Communication Signaling (ICSS)

All servers in CUCM cluster form TCP connecions to each other for ICSS on port 8002- 8004 and keep each other informed

CUCM Publisher holds master copy of Informix database, changes to the database happen on the Publisher and are replicated to subscribers

Each cluster supports one Publisher and up to eight Subscribers. Publisher maintains database and  serves TFTP requests and Subscribers handle phone registration and call control

If Publisher fails, changes can not be made to database, excepting user-facing features such as call forwarding and DND button, etc. Subscriber writes local copy of change and replicates to other subscribers until Publisher returns online

Cisco Unity Connection

Cisco Unity Connection is an enterprise, appliance-based voice-mail solution similar to CUCM

Key features of CUC:

  • Appliance-based: Stable, hardened, appliance-based OS
  • 20,000 mailboxes per server
  • Remote access to voicemail via email, browser, IM and phone
  • LDAP/Active Directory integration
  • Microsoft Exchange supported for calendar integration, text-to-speech, etc
  • Voice Profile for Internet Mail: Standard which allows other voicemail servers to integrate for exchange of voicemail and other messages
  • Active/Active HA cluster with Publisher/Subscriber and Informix DB allows doubling of voicemail ports and mailboxes

If one of HA cluster fails, half the voicemail ports and mailboxes are inaccessible

CUC is able to integrate with other call control systems such as PBX and so does not have close integration with CUCM. CUC is set as an outside system that CUCM communicates with using SIP/SCCP

CUCM to CUC Call Flow:

  1. Incoming voice call hits CUCM from PSTN VG or internal call
  2. CUCM routes call to approriate IP phone
  3. If call is not answered, CUCM forwards call to Voicemail pilot extension
  4. CUCM transfers call to CUC with original extension in SCCP/SIP signaling which CUC uses to find appropriate mailbox
  5. After VM is recorded, CUC calls MWI extension on CUCM to toggle light on IP phone 

All communication takes place using voicemail ports
CUC can also integrate with CME

Cisco Unified Presence

CUP is used to track availability of a user and provide enterprise IM capability

Key features of CUP:

  • Enterprise IM using Jabber XCP
  • Logging functionality for all types of IM communication
  • Can connect to other domains such as Google Talk or WebEx
  • XCP allows CUP to extend to almost any part of the network, for file sharing, app sharing, videoconferencing. XCP integrates with directory services, databases, web
  • CUP application integration supports IPSec or TLS encryption to secure communication

Unified Personal Communicator

Software application that combines softphone, IM client, employee directory, video/web conferencing. Allows tracking of user status and virtual meetings

CUPC uses LDAP for login to the client and a CUPS server on the back end

Sunday, January 11, 2015

CCNA Voice: Traditional Voice Concepts

Traditional Voice to Unified Voice

Analog Voice Terms

Loop Start: Relies on connecting 'tip' and 'ring' wires in an analog device to complete an electrical circuit and causing electrical signal to flow from PSTN CO. Susceptible to Glare

Glare: Caused when a user signals the PSTN CO at the same time that a call is coming in, causing that incoming call to be routed to the user that just picked up the phone

Ground Start: Relies on grounding the analog wires which causes the PSTN to send electrical signal to the device, only used on outgoing calls so this can prevent Glare

Analog Challenges

Signal boosting required as distance increases, boosting also boosts line noise

Separate physical lines required for each phone line causes scaling issues

Digital Voice Terms

Digitizing: Process by which analog vice signals are changed to digital numbers

Time-Division Multiplexing: Allows voice networks to carry multiple conversations at the same time using time-slots for the digitized conversations

T1: Digital circuit comprised of 24 seperate 64-kpbs channels known as DS0, each one of which supports one call. Used in US, Japan, Canada

E1: Digital circuit comprised of 30 seperate 64-kpbs channels known as DS0, each one of which supports one call. Used in areas other than US, Japan or Canada

Channel-Associated Signaling: Binary bits for voice are stolen for signalling, also known as Robbed-Bit Signaling (RBS). Uses eighth bit on every sixth sample in each channel

Common-Channel Signaling: Dedicated T1 channel for signaling information. Also called out-of-band signalling. Most popular method is Q931. For T1 circuits the 24th time-slot is used for signalling, for E1 the 17th time-slot is used for signalling

PSTN Concepts

Analog Telephone: Common device using PSTN, converts audio to electrical signals

Local Loop: Link between customer premises and telecom provider

CO: Provides services on local loop such as signalling, digit collecting, routing calls and call setup/teardown

Trunk: Connection between CO or private switches

Private Switch: Used for business to operate internal PSTN instead of each phone having separate connection to external CO

Digital Phone: Converts audio into digital signal, more efficient than analog

PSTN Numbering Plan

E.164 Numbering Plan was created by ITU and contains:

  • Country Code
  • National Destination Code
  • Subscriber Number
North American Numbering Plan uses:

  • Country Code
  • Area Code
  • CO/Exchange Code
  • Station Code

PBX/Key System Concepts

PBX/Key System: Internally manages phone calls/phones, has several different kinds of cards and equipment. Calls internally are controlled by  PBX/Key System, calls to/from PSTN utilize trunk between PBX/Key System and PSTN CO

Line Cards: Connects telephone handsets to PBX system

Trunk Cards: Connects PBX to PSTN or other PBX Systems

Control Complex: Intelligence behind PBX System, performs call routing, setup/teardown and management functions

VOIP Business Benefits

Reduces cost by allowing use of WAN connections instead of PSTN charges
Reduces cost of cabling, requiring single Ethernet drop
Centralized dial-plan and command/control of calling
Move/Add/Change costs are eliminated
Softphones allow users to use headset and computer as a phone instead of needing hardware
Unified messaging such as email, fax, voice mail
Multiple device ring increases productivity by allowing users to be reached on multiple devices
Feature-rich communication such as screen popup when customer calls into a call center
Compatible standards to allow different vendors to work together

Converting Voice to Data

Average human ear can hear frequencies from 20-20,000 Hz
Human speech uses frequencies from 200-9000 Hz
Telephone channels usually transmit 300-3400 Hz
Nyquist Theorem produces frequencies from 300-4000 Hz

Nyquist Theorem: Accurately reproducing an audio signal require sampling the signal at twice its highest frequency, ie, for a 300-4000 Hz signal to be reproduced would require 8000 samples per second

Quantization: Process by which analog waves are converted into digital signal

1 byte represents value of 0-255, voice scale must be between 127 and -127

Amplitude values common to voice are more tightly spaced

Sampling breaks 8 binary bits in each byte into two components: numeric representor and positive/negative value

G.711 a-law used everywhere except US and Japan, 64kbps

G.711 u-law used in US/Japan, also 64 kbps but sampling valuies are reversed (1 bits are 0 and 0 bits are 1)

Compression measures applied to lower bandwidth requirements

G.729 compresses by sending sample once and instructing device to play that sound for a time value, reduces bandwidth to 8kbps

Mean Opinion Score: Rates quality of voice codecs

G.711 and G.729 are common codecs for all Cisco IP Phones

Digital Signal Processors

DSP: Hardware chip that provides sampling, compression, encoding functions to audio coming into the router

PVDM: Packet Voice DSP Modules, bundle multiple DSPs into one chip

DSP/PVDM can be added directly to the router's motherboard (if supported) or as part of a Network Module

Based on complexity of codec, PVDMs can handle more or less audio calls at once


RTP: Real-Time Transport Protocol, Transport Layer protocol, uses UDP. Provides time stamps and sequence numbers to UDP packet so it can be reassembled in order (sequence) and reduce jitter (time stamp) on remote end. Uses even UDP port between 16384 and 32767 for each audio stream (a two way call will have two one-way RTP streams)

RTCP: Real-Time Transport Control Protocol, Used for statistics reporting. Picks odd number UDP port range between 16384 and 32767. 
  • Packet count
  • Delay
  • Packet Loss
  • Jitter
If RTP stream uses 17654, RTCP will use 17655