Brkuct 3006

New Developments in Unified Communications Design
BRKUCT-3006
Luc Bouchard
2008 Cisco Systems, Inc. All rights reserved.
Cisco Public
Session Scope and Objectives

To explore the various updates available from each of the specialized Unified Communications breakout sessions We will look at highlighted content covering:
Newly available features Newly available architectural or design guidance
The goal is not to replace the need for the actual specialized session it is more to serve as a bulletin board advertising new content.
Cisco Public
TCL/VXML Highlights from US session BRKVVT-2003
Cisco Public
Custom Applications Overview

Custom applications can be deployed on Cisco IOS routers using Tool Command Language (Tcl) and/or VoiceXML
Rich, flexible scripting capabilities integrated in Cisco IOS CPU-intensive features careful platform sizing Application development done by Cisco Developer Support, Cisco partners or end customer
Build custom applications to solve business problems or raise worker productivity

Custom feature development Very powerful applications can be built
Application examples
Prepaid or custom calling services Database checking to support call routing (e.g. do-not-call list) Router-based AA / ACD call queuing
2008 Cisco Systems, Inc. All rights reserved. Cisco Public
Tcl Application Architecture

Tcl script example: Modify Display Name
Match the calling number in a list and modify the display name
Script
PSTN J.B.
H.323/SIP Joe Blow
dial-peer voice 2 pots direct-inward-dial application modify_name
dial-peer voice 1 voip destination-pattern 720555xxxx session-target ipv4:10.0.0.1
Cisco Public
Introducing Tcl Capabilities

Tcl is an interpreted script language, runs locally on router Tcl provides command set, flow control (if, then, else), variables Command set extensions
Cisco extensions: Tcl IVR commands
Cisco Tcl scripts are invoked when a call arrives Ability to run on TDM or IP leg of call
Can be used with pure IP solution, e.g. SIP trunks
Ability to manipulate CALL INFORMATION

E.g. ANI/DNIS, Name Display
Ability to manipulate call before CONNECT

E.g. cut through audio in one direction to get information back to caller; can route a call on more than ANI/DNIS
CALL STATE control

Allow/deny a call to continue, call connect/disconnect, transfer
VoiceXML Application Architecture

VoiceXML Gateway
Cisco IOS voice browser parses and interprets VoiceXML voice dialog HTTP/HTTPS request
VXML
ASR/TTS Server
MRCP Plays audio or streaming prompts and interpret responses (grammars)
PSTN or VOIP
Web Server
Web/app server generates VoiceXML documents defining voice applications presentation logic
User calls via PSTN or VoIP
Presents to, and gathers, user information Web Browser Web Browser Makes client requests to application server and interprets responses to user input Voice Browser Voice Browser
Answers call, fetches VoiceXML docs at URLs
HTML
VoiceXML
ASR, TTS, VoiceXML application server
Maintains application state, dictates Web/Application Web/Application Web/Application Web/Application logic, fetches / stores database Server Server Server Server information, serves responses to client Database Stores and retrieves information
Cisco Public
Database
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Introducing VoiceXML Capabilities

VoiceXML Applications
Markup language for creating voice IVR with DTMF/ASR/TTS Telephone access through voice browser to web content/services Full power of web development and content delivery Browser runs on router webpages downloaded from server
Ability to run on TDM or IP leg of call

Can be used with pure IP solution, e.g. SIP trunks
VoiceXML browser has full VoiceXML 2.0 Compliance

Additionally implements selected VoiceXML 2.1 features
Extensive IVR capabilities: Full control of user dialog flow

Database checks and retrievals Recognition of DTMF key presses, and spoken words/phrases (ASR) Output of synthesized speech (TTS)
Recording of voice phrases or segments

Recording of spoken input; play background beep
Simple call control (e.g. call transfer and disconnect

Application Examples
Service provider:
Prepaid calling card with account number and PIN checks
Contact Center
CME AA and Basic-ACD call queuing Call handling to assist applications like CVP (e.g. survivability)
Customized call routing

Hookflash call transfer on FXO trunks to release hairpinned trunks Receive customized ANI on T1 CAS trunks Time of day call routing and barring Caller-ID-based routing and blocking, or setting call Display Name Time-limited call duration for outbound calls
IVR
Snow-day announcements, and outbound notifications
Non-voice
Router-monitoring/utilization
Tcl and VoiceXML Applications

When to Use What?
Tcl
Standardization Application Type Primary Target Applications Scripting Application location Caller interaction Management Application interaction Industry technology General and Voice Best at call control manipulation and initiating outbound calls Can also do IVR Procedural Scripts, commands, events Local on router Input: DTMF Output: Static/dynamic prompts Largely distributed Largely self-contained
VoiceXML
W3C Standard Voice Best at single user session dialog interaction (IVR) Can also do limited call control Web environment Documents, menus, forms Router and server components Input: DTMF, ASR Output: Sophisticated TTS Centralized management with distributed service delivery Fully web-enabled
Hybrid Tcl and VoiceXML Applications Leverage Best Capabilities of Both

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History and Standards

Tcl
Invented in the late 1980s by John Ousterhout at UC Berkeley, later Sun Microsystems Unix scripting tool, often used for automating test routines based on executing commands and acting on events Tcl Developers Exchange (http://www.tcl.tk/) Cisco IOS Support: Basic Tcl since 11.3(6)NA2 Tcl IVR 2.0 (using Tcl 8.3.4) since 12.1T
VoiceXML
VoiceXML Forum: Founded by industry consortium (AT&T, IBM, Lucent/Avaya, Motorola) W3C voice browser working group (http://www.w3.org/Voice/) Cisco IOS Support: VoiceXML pre-standard 2.0 12.2T VoiceXML 2.0/2.1 12.4.15T
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Script Development and Support

Custom-written scripts
Customers or Cisco partners may develop and support scripts Several script examples posted on Cisco.com as is samples and must be adjusted/tested for a specific application
Cisco Developer Support

Cisco Advanced Services via the Developer Support program develops custom scripts based on a statement of work agreement Cisco Developer Support information can be found on Cisco.com Partners: http://www.cisco.com/web/partners/pr46/tdp/index.html General: http://www.cisco.com/en/US/products/svcs/ps3034/ps5408/ps5418/serv_ho me.html
Cisco TAC support

Supports underlying Tcl/VoiceXML and call state execution in Cisco IOS; supports built-in scripts e.g. CME-ACD and CVP scripts Does not support custom-written Tcl/VoiceXML script logic or troubleshooting Does not write or develop scripts
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Sample Tcl/VoiceXML Scripts on Cisco.com

General Tcl Scripts for IOS
http://www.cisco.com/cgi-bin/tablebuild.pl/tclware
Reference
Tcl Script for CME/SRST

Follow IOS SW download links for CME/SRST, starting at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key
Tcl Programmer Guide sample scripts: Appendix A

http://www.cisco.com/en/US/products/sw/voicesw/ps2192/products_programmi ng_reference_guide_chapter09186a00801ce92e.html
Embedded Event manager (EEM) Tcl Scripts

http://forums.cisco.com/eforum/servlet/EEM?page=main
Cisco Developer Support Tcl and VoiceXML Sample Scripts

http://www.cisco.com/en/US/products/svcs/ps3034/ps5408/ps5418/serv_home. html
Cisco Public
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Voice at the Edge, or Core

Centralized Model
Cisco Voice Gateway IVR Server
VoIP
Please enter your account number
Presentation logic executes on centralized IVR server Media streams terminate on IVR server No state awareness at gateway
Distributed Model
Cisco Voice Gateway
VoIP
VXML doc
Web Server
Please enter your account number
Presentation logic distributed to edge of network IVR executes on Gateway Media streams terminate on Gateway State awareness in gateway
Scaling, performance and cost benefits

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Tcl
Small-Medium Business: Basic ACD

Small offices often require some call queuing and distribution functionality Cisco Unified CME Basic-ACD queuing and AA functionality is implemented in Tcl
AA announcement, call queuing and agent availability monitoring, as well as extending a call to an available agent
2
AA
3 1 Script
Agent Group 1 Agent Group 2
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Tcl
Enterprise: FXO Hookflash Transfer

Example: Car rental office locations Small branch offices have 2-4 FXO trunks Transferring a call from AA to main Customer Service toll-free number ties up 2 FXO trunks
Tcl script in branch does FXO hookflash on the FXO trunk to transfer the call back to the PSTN and to release the trunks
HQ Contact Center
Script 2
FXO
CUE AA
Small Branch Office

Note: Hookflash Requires Tcl (not VoiceXML), Only Platforms with Analog Ports Support this Capability
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VoiceXML
Enterprise: Do-Not-Call List

Enhance the productivity of employees who make outbound calls to potential customers VoiceXML application on PSTN gateway intercepts outbound call and checks against a database of do-not-call telephone numbers VoiceXML application allows or denies the call to proceed
VoiceXML Application Server
3
Do-not-call database
2
Script
IP
1
CUCM Employee initiates outbound calls

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Cisco Public
Hybrid
Enterprise: Distributed Contact Center

Provides access to centralized IVR applications via branch office Provides edge queuing/IVR for contact center solutions keeps RTP off WAN until agent is selected
Supported with H.323 and SIP (CVP 4.0)
Call survivability at PSTN Gateway provided by Tcl script

Active Voice Call (speechpath, RTP) GW Control Via HTTP/VoiceXML
CVP
VoiceXML RTP
IP
When Agent Available, Active Voice Call Rerouted to Agent
Cisco Public
Database
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Tcl
Service Provider: Prepaid Calling Card

DGK
PSTN
AS5000 Gateways GK GK
AS5000XM
SBC
RADIUS
TFTP TFTP Server for Script and Prompt Storage Server Hosting Prepaid Calling Card Database
PSTN
Allows SP to offer a demarcation point for prepaid SIP/H.323 interconnects Maintains investment in current billing infrastructure Cisco AS5000XM includes SBC functionality and TDM PSTN Gateway functionality in the same chassis RADIUS/AAA integration for user authentication
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Gateways Highlights from US session BRKVVT2010
Cisco Public
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High-Density Packet Voice/Fax Feature Card (AS5X-FC) and DSP Module (AS5X-PVDM2-64) Cisco 5000 Series Gateway DSP Cards
High-Density Packet Voice/Fax Feature Card AS5X-FC supports from 1 to 6 AS5X-PVDM2-64 modules
AS5XPVDM2-64 Channels / Module 64 32 24 AS5X-FC Channels / Card 384 192 144
Codec Type
Low Complexity
le odu M SP D 2-64 ceable DM -PV Repla X AS5 Field

Medium Complexity High Complexity
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H.320 ISDN Video Connectivity

CUCM
IP
H.323
H.3
PSTN SCCP
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Video Voice
H.320 to/from SCCP and H.323 Audio/Video Endpoints Supported in SRST, also with PRI and BRI Up to 16 channels of H.320 video (1M video calls) DSPs shared between video and voice Interoperability with non-Cisco IP endpoints (Polycom/Tandberg) in a future release (p*64 ISO-13871 mode) RFC 3550 (all Audio), RFC 2032 (H.261), RFC 2190 (H.263), RCF 2429 (H.263+), ITU-T H.241 Annex-A (H.264)
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Integrated Voice + Data Access

Single T1/E1 Voice Single T1/E1
PSTN
WAN
Voice
Data
Data
North-America: Serial Data

Voice: ds0 or pri-group Data: channel-group (HDLC, FR, PPP) Timeslot allocation: static H.323 and SIP only Has always been supported Data (Channel-Group: FR, HDLC, PPP, MLPPP) Voice (ds0, or pri-group) D-Channel (if pri-group is Used), Controls Voice Channels Only
Europe/Aus: PRI Data

Voice + data: pri-group Data: Dialer i/f using PRI channels Timeslot allocation: dynamic H.323 and SIP only Supported as of 12.4.9T Voice + Data (pri-group)
Voice channels:
Voice bearer CAP
T1
T1/E1
Data channels:
64K unrestricted bearer CAP
D-Channel Controls All Channels
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Integrated Voice + Data PRI

Dial Backup Via PSTN Connection During WAN Failure
Single T1/E1 PRI Voice Single T1/E1 PRI Voice
PSTN
Backup Data
Data
WAN
Data
PRI WAN Connection Sharing Same Interface as Voice (SP Offering)

Single T1/E1 PRI Voice Data
Data Services Voice Services
Single T1/E1 PRI Voice Data
Integrated Service Provider

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Terminating Voice, Video + Data on a Router

Voice and video traffic is terminated on DSPs Data traffic is terminated on HDLC controllers
Channel-group: need 1 HDLC controller per group PRI: need 1 HDLC controller per channel (timeslot) dial-peer voice 3 pots destination-pattern 5556 information-type video port 1/0/0:15 interface Serial1/0/0:15 isdn integrate calltype all
Requires 12.4.9T for data, and 12.4.11T for video
ISR Voice Gateway

VoIP Packets
DSPs
TDM (Voice/Video)VWIC TDM (Data)
IP
Cisco IOS SW Data Packets
T1/E1
TDM Backplane
HDLC Controllers
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Real-Time Monitoring Tool Trace and Log Central
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Overview
Formerly known as Trace Collection Tool Used for review and management of trace and log files Enhanced capabilities beyond what CLI offers
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System Tree
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Product Grouping
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Unity Connection Log Viewer
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External Trace Viewer

Associate trace type with internal and external viewers Remote Browse supports right-click on trace to associate trace file with viewer Local Browse utilizes user-configured viewer association
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CODEC and CAC Considerations
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Call Admission Control codec and regions considerations

These are not codec choices, but rather bandwidth maxima!
region SF NY SF 711 729 NY 729 711
For any call, UCM will use the highest-bandwidth codec mutually supported by the endpoints, up to the maximum allowed by the Regions configuration If there are two mutually supported codecs that fit within the maximum, a non-modifiable preference is used In UCM, when both endpoints support 711 and 722, and the Regions config. allows 711 (64kbps, really), 722 is chosen
722 call 729 call
IP WAN
Phone A 2345 San Francisco Location: SF Region: SF

Phone B 1235
Phone C 1236
PSTN
711 call
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New York
New York Location: NY Region: NY

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Communications Manager 6.0 Database Resiliency
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Overview
There have been a number of changes in the Communications Manager 6.0 database architecture. Most significant are the ability to write and read to the local database during normal operation. In addition, change notification is now done locally. This improves database performance and reliability.
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Failover and Redundancy

Database Resiliency (UCM Version 6.0)
Publisher (all data writable)
UC Manager Cluster
Informix Dynamic Server (IDS) Replication
IDS
IDS
IDS
User Facing Features:

Call Forward All Message Waiting Indicator ( MWI) Privacy Enable/Disable Device Mobility Extension Mobility Login/Logout Do Not Disturb Enable/Disable Hunt Group Login/Logout CTI CAPF status for end user Credential hacking & authentication
IDS
IDS
Informix Dynamic Server (IDS) Subscribers (User facing features Writable)
Bidirectional User facing feature replication Logically Unidirectional DB replication from Publisher
Cisco Public
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Communications Manager 6.0 Presence UCM 6.0 SRND, US session TECVVT-2001
Cisco Public
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Overview
Cisco Unified Presence takes advantage of Session Initiation Protocol (SIP) technology to support new voice services in the enterprise environment.
Cisco Public
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Whats New Guide

Popup displayed on CUP CCMAdmin List of new features and links to CCO documents Able to toggle off, but always under Help menu
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CCMAdmin Enhancements
CUPC Client Version
Located on CUPC User Settings page Displays CUPC client version on a per-user basis Version displayed is from the most recent login
CM Navigation Link
Located on main CUP CCMAdmin Links CUP directly to its corresponding CUCM Publisher Future CUCM versions will support navigational link back to CUP
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Privacy
Customizable for each user Default policy applies to all watchers ACLs can be added to create granular lists of watchers Custom device types may be defined for policies Select what presence watchers are allowed to see Precedence-based rules for determining reachability Hide Presence Information for specific devices Filtering rules applied before reachability rules
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CM 6.0 Alignment
PUBLISH of presence data from CM to CUP Line Appearance-based presence Do Not Disturb for phones affects presence Mobile phone presence IPPM contact details support line appearance-based presence
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Sametime Click-to-Call
Sametime can integrate with CM to provide Click-toCall in the Sametime client Required software
Sametime Connect 7.5 Domino Server Directory Server Sametime Server with Telephony Conference Service Provider Interface (TCSPI) and Click-to-Call plug-in Cisco Unified Communications Manager
No configuration in Sametime Connect
Cisco Public
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TCSPI Plug-in
Plug-in extends TCSPI Java classes and interfaces. CM acts as 3rd party call controller for setting up the call between caller and callee Plug-in sends out of Dialog SIP REFER to CM with callee phone number in Refer-To header CM will first initiate a call to callers phone When caller picks up, call will be initiated to callees phone CM will send NOTIFY with state of the dialog in body of the request to plug-in Plug-in will use the notification to update the chat window with state of the call
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Interclustering
Previous versions of CUP require all buddies reside within the local CM cluster
No presence status for non-local buddies Attempting to send an IM to a non-local user would fail
CUP 6.0 supports intercluster buddies

Users may add buddies that reside anywhere in the domain, regardless of cluster CUP server will track down buddies in any cluster, and provide presence status to the subscriber CUP server will forward IMs from the local cluster to the recipient anywhere in the domain
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Exchange Calendar Integration

Incorporates Exchange calendar information into Presence availability User can choose to include the status of meetings Rules impact calendar-based presence status Least available if multiple concurrent meetings
Away if any of the current meetings, has a status of out-ofoffice Busy if any of the current meeting has a status of busy, but not away Free if all of the meetings are either free or tentative
Cisco Public
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Exchange Calendar Integration (Cont.)

PE is configured to connect to Microsoft Exchange PE connects to Exchange using WebDAV protocol PE sends WebDAV subscription for CUP users PE queries for changes for next hour Exchange notifies CUP when calendar has changed and PE queries for changes PE sets up timers to signal the transitions indicated in the users calendar
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Meeting Notification
Automatic just-in-time Exchange meeting reminders to IPPM Join softkey to allow IPPM users to join MeetingPlace Todays meetings IPPM softkey to display days meetings Dial softkey to allow IPPM users to dial the bridge Roster softkey to allow IPPM users to see participant list Reminder or IM message can be sent to a participant selected or all the participants on roster CallBack softkey for automatic callback to the phone when a Meeting Place meeting starts
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LLDP, LLDP-MED and CDP support
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CDP, LLDP and LLDP-MED

Cisco Discovery Protocol (CDP) is Cisco-proprietary, and runs when two Cisco devices are directly connected to each other. CDP v2.0 is extended to support endpoint-specific information Link Layer Discovery Protocol (LLDP) is a standards-based protocol covering the basic device discovery functionalities Link Layer Discovery Protocol -Media Endpoint Devices (LLDPMED) is an extension of LLDP supporting endpoint-specific information A complete comparative description is available at: http://cisco.com/en/US/partner/tech/tk652/tk701/technologies_white_ paper0900aecd804cd46d.shtml
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Protocols allowing directly connected devices to discover information about each other CDP (e.g.: VLAN, Ext. Trust value, power)
Catalyst Switch
LLDP -
MD (e.g.: VLAN, power) E
3rd party Switch
3rd party phone
LLDP -
MD (e.g.: VLAN, power) E
Catalyst Switch
3rd party switch
LLDP (e.g.: chassis ID, port ID, capabilities, inventory management )
Catalyst Switch
Cisco Public
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Cisco phones supporting LLDP include:
7906G, 7911G, 7931G, 7941G/7941G-GE, 7942G, 7945G, 7961G/7961G-GE, 7962G, 7965G, 7970G/ 7971G-GE, 7975G with s/w version 8.3(3) or newer.
Cisco Switches supporting LLDP-MED include:

Catalyst 3750, 3560, 3750-E, 3560-E series and 2960, with IOS version 12.2(37)SE or newer. Cat 4K is coming (ETA: Q1CY08)
Cisco Public
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UCMNew Features Highlights from UCM 6.0 SRND, US session TECVVT2000
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Deployment Models
Centralized Call Processing
Applications (VMail, IPCC, MP) CUCM Cluster SRST/CME-Enabled Router
PSTN
IP WAN
Headquarters
CUCM cluster at central/HQ site Applications and DSP resources can be centralized or distributed Supports up to 30,000 phones per cluster If WAN is busy, transparent use of PSTN (AAR)
See BRKVVT-3000 Advanced Dial Plan Design for details
Branch A
SRST/CME-Enabled Router
Branch B
Survivable remote site telephony for remote branches Maximum 1000 branches per cluster (500 branches before CUCM 6.0)
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Unity/Unity Connection Cisco Unified Phone ViewVisual Voicemail

Phone user accesses mailbox and reviews messages visually through visual message locator functionality. User has the ability to select a message, take actions, view message header details, or jump to a specific message in the stack User dials into
voice mail system, enters password, and presses
5
JKL
Unity/Unity Connection
HTTP
Cisco Public
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Unified MeetingPlace/MeetingPlace Express

Integrations
Access from Multiple Devices and Applications to Better Integrate Conferencing into your business Processes
Calendar, E-mail
Notes, Outlook, SMTP e-mail
Web Browser
IM, Communications Client

CUPC, MOC, Jabber, SameTime
Mobile Client
Unified Mobile Communicator
Cisco IP Phone
Any Touchtone Phone

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Unified MeetingPlace
MeetingPlace for MOC/LCS (OCS)
Improve Collaboration and Speed Business Process by Leveraging Presence and IM for Fast and Easy Ad-Hoc Conferencing
Directly from Office Communicator (MOC):

Leverage presence to determine availability Initiate ad-hoc audio conference Invite users and connect (outdial) to conference Control voice conference (e.g. Mute, eject, lock, end conference)
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Unified MeetingPlace/MeetingPlace Express

Cisco Unified Phone ViewVisual Conferencing
Allows phone user to setup and attend MeetingPlace meetings from phone interface Active conference roster and current speaker indication
User presses Services button, selects Visual Conferencing service and logs on using username and PIN
MeetingPlace/ MeetingPlace Express
HTTP
Cisco Public
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Mobility Highlights from UCM 6.0 SRND, TECUCT-1001, BRKUCT-3007, US session BRKVVT-2017
Cisco Public
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Device Mobility Considerations

High-Level BehaviorUCM 4.2 and 6.0 Only! Determines that the device has moved to new location based on the devices IP subnet Dynamically associates roaming device pool to devices that move to a different site Message displayed on phone screen for a few seconds when it registers with UC Manager:
Device in Home Location Device in Roaming Location
Cisco Public
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Device Mobility
Device Pool Changes
Device Pool
UC Manager Group Auto-reg CSS
Roaming Sensitive Settings
Common Profile (new)

Softkey Template Network Hold MoH Audio Source User Hold MoH Audio Source MLPP Indication MLPP Preemption MLPP Domain
Impacts CAC, Media Resource & SRST
Impacts Dial Plan
Date/Time Group Region MRGL Network Locale SRST Reference Location Softkey Template Physical Location Network Hold MoH Audio Source Device Mobility Group User Hold MoH Audio Source MLPP Indication Device Mobility Related Information MLPP Preemption MLPP Domain Device CSS AAR Group AAR CSS
Device
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Device Mobility
New Concepts
Device Mobility Info
SJCV1_dmi
10.1.1.0/24
Device Pool
Physical Location
Device Mobility Group
SJC-A_dp
(building A)
SJCD1_dmi
20.1.1.0/24
SJC-B1_dp
(building B)
SJC_phyloc
(SJC campus)
SJCD2_dmi
20.1.2.0/24
SJC-B2_dp
(building B)
US_dmg
RTPV1_dmi
10.2.1.0/24
RTPD1_dmi
10.2.1.0/24
RTP_dp
RTP_phyloc
(RTP campus)
LONV1_dmi
10.42.1.0/24
LOND1_dmi
20.42.1.0/24
LON_dp
LON_phyloc
(LON campus)
EUR_dmg
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The Big Idea Is to Track Phones Based on Subnets
voice subnet: 10.1.1.0/24 data subnet: 20.1.1.0/24 data subnet: 20.1.2.0/24 voice subnet: 10.2.1.0/24 data subnet: 20.2.1.0/24 voice subnet: 10.42.1.0/24 data subnet: 20.42.1.0/24
1000
2000
3000
SJC
RTP
LHR
Note: When roaming from SJC Note: When roaming from SJC to LHR, we are crossing DMGs to LHR, we are crossing DMGs Dial Plan-related information Dial Plan-related information does not change. does not change.
Roaming Sensitive Settings Change when roaming between physical locations. DMG not a factor. Device Mobility Related Information Changes only when roaming within the same DMG.
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RTP Mobile User at Home Location
RTP
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RTP Mobile User at SJC Roaming Location
SJC
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Campus Mobility
Dual-Mode Phones
PSTN Gateway
Enterprise
Cisco Unified Presence
GSM/ Mobile Network
PSTN
Cisco Unified CM cluster
MP MP
Unified MeetingPlace/ MeetingPlace Express
Unity/ Unity Connection
WLAN
(802.11b/g)
Nokia
WLAN AP
Active Directory
Dual-mode phones provide the ability to use either PSTN/GSM or WLAN connectivity for making and receiving calls.
When on the WLAN, the mobile phone uses SCCP or SIP Cisco client to register with CUCM as a phone. When the WLAN is unavailable, the mobile phone uses PSTN/GSM for calls. Manual handoff of calls between the PSTN/GSM and WLAN network is possible
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Unified Mobility
Configuration and Call Routing Concept
1 RD Profile per Mobility User
Remote Destination Profile

Virtual Phone
IP Phone
Line Level Configuration
DN: 408 555-1234 (Partition/Calling Search Space)
Shared Line
DN: 408 555-1234 (Partition/Calling Search Space)
Line Level Configuration
RD Profile Level Configuration: Device Pool Calling Search Space Rerouting Calling Search Space User/Network Hold Audio Source Call routing and MoH behavior for Remote Destination devices
Device Level Configuration: Device Pool Common Device Configuration Calling Search Space Media Resource Group List User/Network Hold Audio Source Call routing and MoH behavior for IP Phone
408 555-7890
Remote Destination #1
408 555-6789
Remote Destination #2
XXX XXX-XXXX
Remote Destination N
Configuration per Remote Destination basis: Timers (Answer Too Soon/Late and Delay Before Ringing) Allowed/Blocked Access Lists 67
Cisco Public
Unified Mobility
Mobile Connect
Call rings Remote Destination 408 555-7890
Enterprise
Call to Remote Destination 408 555-7890 routed via gateway
408 555-7890
6 PSTN Gateway
PSTN
Call placed to associated Remote Destination 408 555-7890
Remote Destination: 408 555-7890
1
Dials: 1 408 555-1234 Phone A
4 Remote Destination Profile
2
Call extended to Remote Destination Profile
Call extended to desk phone
Mobile Users Enterprise DN
DN: 408 555-1234
DN: 408 555-1234
Shared Line
Call to Mobile Users Enterprise Directory Number Rings at Desk Phone and Remote Destination Phone: Note: No changes are required on
Call can be answered at either phone Once answered all other call legs are cleared
mobility users Remote Destination phone

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Unified Mobility
Remote Destination Pickup
Remote Destination and Desk Phone Pickup

Call rings remote phone. Once answered call continues uninterrupted between caller and remote phone User presses Mobility softkey and then Select softkey to pickup on remote destination phone
5551234
MobileConnect On
3
408 555-7890
PSTN Gateway
jsmith
PSTN
DN: 408 555-1234
1
Mobile Connect call answered and in progress at desk phone
Enterprise Desk Pone Pickup

Mobile Connect call answered and in progress at remote destination
Phone A
Once Mobile Connect call is in progress there are two types of pickup: 1. Remote Destination Pickup Mobile user can pickup inprogress desk phone call at Remote Destination phone Desk Phone Pickup Mobile user can pickup inprogress remote phone call at desk phone
1 PSTN Gateway
408 555-7890
Upon remote destination hang up (or mid-call hold) user can pickup at desk phone by pressing the Resume softkey
PSTN
3
Call continues uninterrupted between caller and desk phone 69
DN: 408 555-1234
2.
Enterprise
Phone A
Unified Mobility:
Mobile Voice Access
Dials: 1 408 555-2345
Enterprise
1 2
User enters ID, PIN#, and navigates menus via keypad. Enters 1 to make call and then enters 9 1 972 555 3456
408 555-7890
H.323 VXML PSTN Gateway
Unified CM sends IVR prompts to VXML H.323 gateway and gateway collects and forwards user input
PSTN
Mobile Voice Access DID: 408 555-2345
6 5
972 555-3456
Call rings PSTN number 972 555-3456 Call to PSTN number 972 555-3456 routed via gateway
Call originated via users Remote Destination Profile
Mobility User Can Use Communications Infrastructure to Make Enterprise Calls from Their Remote Destination Phone.

DN: 408 555-1234
Shared Line
DN: 408 555-1234
Call made to Enterprise Mobile Voice Access number User follows IVR prompts and enters information to make call User can also disable and enable Mobile Connect on a per remote destination basis
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Unified Mobility
Off-Box Mobility with MobilityManager (CUCM 4.X/5.X)
408 555-7890
Call rings Remote Destination 408 555-7890 Call to Remote Destination 408 555-7890 routed via gateway
Enterprise
Mobility Manager Route Point
Call routed via MobilityManager Route Point
8 PSTN Gateway
PSTN
Cisco Unified MobilityManager

Call placed to Remote Destination 408 555-7890 on Outgoing CTI port
Outgoing CTI Port
1
Dials: 1 408 555-1234 Phone A
Call extended to desk phone
Call offered to Unified MobilityManager Application Server
Prior to CUCM 6.0, Mobility Functionality Is Provided via Cisco Unified MobilityManager
Shared Line CTI Port
Call extended to Shared line CTI port
DN: 408 555-1234
DN: 408 555-1234
Shared Line
CTI is required for interaction between MobilityManager application server and Cisco Unified CM, but behavior is the same.
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Unified Mobility
Mid-Call Features
Remote Destination phone invokes mid-call features by sending DTMF digits to Unified CM.
Enterprise
In-progress call can be transferred to another phone or additional phones can be conferenced in
408 555-7890
In-progress MobileConnect call between Remote Destination 408 555-7890 and Phone A
2 PSTN Gateway 1 4
PSTN
DN: 408 555-2345
Phone A
MoH is forwarded from Unified CM to Phone A when mid-call Hold, Transfer, and Conference features are invoked

DN: 408 555-1234 DN: 408 555-1234
Once a Mobile Connect or Mobile Voice Access call is in-progress at mobile users Remote Destination phone:
Shared Line
Mid-call features like Hold, Transfer, and Conference can be invoked via Smart Phone softkeys or manual key presses. DTMF tones are sent from the Remote Destination phone to the CUCM via the Enterprise PSTN gateway.
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Unified Mobility
2 Stage Dialing
Remote Destination phone invokes two-stage dialing by dialing pre-configured Enterprise Feature Access DID 408 555-2345 Once call is connected Remote Destination phone sends DTMF digits to Unified CM: <PIN_number> # 1 # 9 1 972 555 3456
Enterprise
1
408 555-7890

Enterprise Feature Access DID: 408 555-2345 Call originated via users Remote Destination Profile
PSTN Gateway
PSTN
972 555-3456
4
Call to PSTN number 972 555-3456 routed via gateway
2 Stage Dialing provides identical functionality to Mobile Voice Access but without the IVR
3 Remote Destination Profile

DN: 408 555-1234 DN: 408 555-1234
5
Call rings PSTN number 972 555-3456
Shared Line
Call made to Enterprise Feature Access number User presses Smart Phone softkeys or manually keys digits (sent via DTMF) to make call User can also disable and enable Mobile Connect on a per remote destination basis
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Unified Mobility Communicator

Architecture
CUMC Smart Client provides access to many Enterprise applications: Conferencing, Presence, and Voicemail PSTN Gateway
Enterprise
Cisco Unified CM Cluster
MeetingPlace Express
PSTN
MPE
DMZ
SSL over TCP/IP
JTAPI
SSL over TCP/IP
Unity WebDAV Exchange LDAP
Cisco Unified Mobile Communicator
Internet
CUMA Proxy CUMA Enterprise
Cisco Unified Mobile Communicator is a Mobility solution that allows mobile users to experience Enterprise telephony on their mobile phones using a Smart Phone client
Active Directory
Cisco Unified Mobility Advantage
Takes advantage of Unified Mobility (Mobile Connect, etc.) to provide Enterprise features. Data from mobile phone backhauled over SSL TCP/IP to CUMA server in Enterprise (requires mobile data plan) Various handsets and mobile operating systems supported: Nokia, Motorola, Blackberry, LG, Audiovox, Samsung, Symbian, BREW, Blackberry
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Dial Plan Considerations for Mobility Highlights from UCM 6.0 SRND, BRKUCT-3007
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Mobility: Dial Plan Implications

1Without Mobility
B- IP phone 8 555 1001 calls 8 555 1000 (**) Called: 8 555 1000
Calling: 8 555 1001
(**) destination must be found in calling phones line + device CSSes. e.g.: in our case: L_1001,D_1001. DN: 8 555 1000 DID: 408 555 1000 Line CSS: L_1000 Device CSS: D_1000
Ca Ca lling: lled 51 :8 45 55 55 5 1 12 00 34 1
DN: 8 555 1001 DID: 408 555 1001 Line CSS: L_1001 Device CSS: D_1001
Calling: 514 555 1234 Called: 408 555 1001
(*)
(*) destination must be found in GWs CSS. Use GW config or translation patterns to adapt called number to enterprise addressing scheme e.g.: in our case, adapt 408 555 1001 to 8 555 1001
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514 555 1234
PSTN
CSS: GW_CSS
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A- mobile phone calls 408 555 1001


2New Configuration
DN: 8 555 1000 DID: 408 555 1000 Line CSS: L_1000 Device CSS: D_1000 Remote Destination Profile CSS: css_mrk_1 Reroute CSS: css_mrk_2 Calling party transformation css: css_mrk_3 Remote Destination Number: 5145551234
New configuration is tied to the DN of the phone
PSTN
514 555 1234 CSS: GW_CSS
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RDP and Remote Destination Number Associated to DN
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RDP and Remote Destination Number Associated to DN
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3With Mobility: Mobile Connect
IP Phone 8 555 1001 Calls 8 555 1000
DN: 8 555 1000 DID: 408 555 1000 Line CSS: L_1000 Device CSS: D_1000 Remote Destination Profile CSS: css_mrk_1 Reroute CSS: css_mrk_2 Calling Party Transformation CSS: css_mrk_3 Remote Destination Number: 5145551234 Calling: 8 555 1001, Called: 5145551234 (*)
Calling: 85551001 (**) Called: 51455512 34 (***)
DN: 8 555 1001 DID: 408 555 1001 Line CSS: L_1001 Device CSS: D_1001 (*) destination must be found in RDPs rerouting CSS e.g.: in our case, css_mrk2 (**) calling party number may be changed my Calling Party Transformation patterns contained in the Calling Party Transformation CSS e.g.: in our case, css_mrk3 (***) called party number can be changed by application dial rules Note these are not contained in a CSS.
PSTN
514 555 1234 CSS: GW_CSS
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3With Mobility: Transformation Patterns
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3With Mobility: Application Dial Rules
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4With Mobility: Mobile Connect Enhanced
IP Phone 8 555 1001 Calls 8 555 1000
DN: 8 555 1000 DID: 408 555 1000 Line CSS: L_1000 Device CSS: D_1000 Remote Destination Profile CSS: css_mrk_1 Reroute CSS: css_mrk_2 Calling party transformation css: css_mrk_3 Remote Destination Number: 5145551234 Calling: 8 555 1001, Called: 5145551234 (*)
Calling: 40855510 01 (**) Called: 91514555 1234 (***)
DN: 8 555 1001 DID: 408 555 1001 Line CSS: L_1001 Device CSS: D_1001 (*) destination must be found in RDPs rerouting CSS e.g.: in our case, css_mrk2 (**) calling party number may be changed my Calling Party Transformation patterns contained in the Calling Party Transformation CSS e.g.: in our case, css_mrk3 (***) called party number can be changed by application dial rules Note these are not contained in a CSS.
PSTN
514 555 1234
CSS: GW_CSS
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5With Mobility: Inbound Calls
DN: 8 555 1000 DID: 408 555 1000 Line CSS: L_1000 Device CSS: D_1000 Remote Destination Profile CSS: css_mrk_1 Reroute CSS: css_mrk_2 Calling party transformation css: css_mrk_3 Remote Destination Number: 5145551234
Calling: 514 555 1234 Called: 408 555 1001
(*)
PSTN
514 555 1234 CSS: GW_CSS
A- Mobile Phone Calls 408 555 1001

Ca Ca lling: lled 85 : 8 551 55 0 5 1 00 00 1
(*) if Calling Party Number matches a remote destination number from any RDP in the cluster, the calling party number is replaced with the RDPs associated DN AND the destination must be found in the concatenated Line CSS and RDP CSS e.g.: in our case, L_1000,css_mrk_1
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QoS Highlights from US Session TECRST-2500
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Cisco Catalyst 4500 (Sup II+ Through Sup V) and 4948

QoS Design Considerations
QoS disabled by default Full DSCP-range is supported Classification can be done by trust states, standard and advanced IP ACLs, or MAC ACLs No mls prefix in command syntax Policing rates can use k, m, or g for kbps, mbps, or gbps Supports per-port/per-VLAN policing SupV-10GE supports User-Based Rate Limiting (UBRL) Minimum policing granularity is 8 kbps Supports 4Q1T queuing or 1P3Q1T queuing
Q3 can be configured as a priority queue DSCP values can be mapped to queues Supports bandwidth allocation and shaping on certain linecards Congestion avoidance algorithm is Dynamic Buffer Limiting (DBL)
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Cisco Catalyst 4500 QoS Design

IP Phone + PC + Scavenger(Basic) Model Example: Part One
CAT4500-SUP4(config)#qos map cos 5 to dscp 46 ! Modifies CoS-to-DSCP mapping to map CoS 5 to DSCP EF CAT4500-SUP4(config)#qos map dscp policed 0 24 to dscp 8 ! Excess DVLAN & VVLAN traffic will be marked down to Scavenger (CS1) CAT4500-SUP4(config)#class-map match-all VVLAN-VOICE CAT4500-SUP4(config-cmap)# match ip dscp ef CAT4500-SUP4(config-cmap)# CAT4500-SUP4(config-cmap)#class-map match-all VVLAN-CALL-SIGNALING CAT4500-SUP4(config-cmap)# match ip dscp cs3 CAT4500-SUP4(config-cmap)#policy-map IPPHONE+PC-BASIC-VVLAN CAT4500-SUP4(config-pmap)#class VVLAN-VOICE CAT4500-SUP4(config-pmap-c)# police 128 kbps 8000 byte exceed-action drop ! Only one voice call is permitted per switchport VVLAN CAT4500-SUP4(config-pmap-c)#class VVLAN-CALL-SIGNALING CAT4500-SUP4(config-pmap-c)# police 32 kbps 8000 byte exceed-action policed-dscp-transmit ! Out-of-profile Call-Signaling is marked down to Scavenger (CS1) CAT4500-SUP4(config-pmap-c)#class class-default CAT4500-SUP4(config-pmap-c)# set ip dscp 0 CAT4500-SUP4(config-pmap-c)# police 32 kbps 8000 byte exceed-action policed-dscp-transmit ! Unauthorized VVLAN traffic is marked down to Scavenger (CS1)
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Cisco Catalyst 4500 QoS Design

IP Phone + PC + Scavenger(Basic) Model Example: Part Two
CAT4500-SUP4(config-cmap)#policy-map IPPHONE+PC-BASIC-DVLAN CAT4500-SUP4(config-pmap-c)#class class-default CAT4500-SUP4(config-pmap-c)# set ip dscp 0 CAT4500-SUP4(config-pmap-c)# police 5 mbps 8000 byte exceed-action policed-dscp-transmit ! Out-of-profile data traffic is marked down to Scavenger (CS1) CAT4500-SUP4(config-pmap-c)# exit CAT4500-SUP4(config-pmap)#exit CAT4500-SUP4(config)# CAT4500-SUP4(config)#interface FastEthernet2/1 CAT4500-SUP4(config-if)# switchport voice vlan 110 CAT4500-SUP4(config-if)# switchport access vlan 10 CAT4500-SUP4(config-if)# qos trust device cisco-phone
! VVLAN ! DVLAN ! Conditional Trust
CAT4500-SUP4(config-if)# vlan-range 110 CAT4500-SUP4(config-if-vlan-range)# service-policy input IPPHONE+PC-BASIC-VVLAN ! Applies VVLAN policers to VVLAN (110) via Per-Port/Per-VLAN policing CAT4500-SUP4(config-if)# vlan-range 10 CAT4500-SUP4(config-if-vlan-range)# service-policy input IPPHONE+PC-BASIC-DVLAN ! Applies DVLAN policer to DVLAN (10) via Per-Port/Per-VLAN policing CAT4500-SUP4(config-if)#end CAT4500-SUP4#
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Cisco Catalyst 6500

AutoQoS VoIP (CatOS Only)
set qos autoqos set port qos 3/1 autoqos voip ciscoipphone
set qos enable set qos map 2q2t tx 2 1 cos 1 set qos map 2q2t tx 2 1 cos 2 set qos map 2q2t tx 2 1 cos 3 set qos map 2q2t tx 2 2 cos 5 set qos drop-threshold 2q2t tx queue 1 100 100 set qos cos-dscp-map 0 10 18 26 34 46 48 56 set qos ipprec-dscp-map 0 10 18 26 34 46 48 56 set qos policed-dscp-map 0,26,46:0 set qos policed-dscp-map 1:1 set qos policed-dscp-map 63:63 clear qos acl all #ACL_IP-PHONES set qos acl ip ACL_IP-PHONES trust-cos ip any any # commit qos acl all ! set vlan 100 3/1 set port qos 3/1 trust-device ciscoipphone set trunk 3/1 off negotiate 1-1005,1025-4094 set spantree portfast 3/1 enable set port qos 3/1 trust trust-cos set qos acl map ACL_IP-PHONES 3/1 set port qos 3/1-48 policy-source local set port channel 3/1 mode off
Options:
autoqos voip cisco-phone autoqos voip ciscosoftphone auto qos voip trust
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NBAR on Supervisor 32 PISA

Network Based Application Recognition
NBAR Policy can mark HTTP NBAR Policy can mark HTTP data as high priority and rate data as high priority and rate limit both E-Donkey and limit both E-Donkey and Netshow traffic ensuring Netshow traffic ensuring priority for internal HTTP traffic priority for internal HTTP traffic
Link Utilization
E-Donkey Netshow HTTP E-mail 60% 30% 5% 25%
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Network Based Application Recognition

IP packet TCP/UDP packet Data packet
ToS Protocol Source IP addr Dest IP addr Src port
Dst port
Sub-port/deep inspection
Full-packet, stateful inspection identifies traffic type Protocol discovery analyzes multi-packet behavior and application signatures Enables application of QoS policies to traffic flows Identifies over 90 applications and protocols
Peer to Peer: E-Donkey, Bit-Torrent, SKYPE Protocol Traffic: RSTP, SIP, L2TP, MPLS to IP Corporate Application Traffic: CITRIX ICA, SAP
Support dynamic protocol definition language module (PDLM) upload for new protocols
Stateful, deep packet inspection Tracks flows based on static or random ports and the Layer 7 payload signature Can recognize applications on non-default ports for protocols
Supports User defined applications and sub-port classification

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FPM on Supervisor 32 PISA

Flexible Packet Matching
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Defining the FPM Rule

Any field defined in the Protocol Header Definition (PHDF) files can be matched Supports pattern matching through regular expressions and string matching: Up to 48 regular expressions with 32 Bytes windows in phase 1.0 Complex rules are more expensive, so best is to configure regular expressions for a given type of packet (stack)
Slammer
class-map type access-control match-all
slammer
IP
Gnutella
UDP
Payload
Stack
description match field match field match start
"match on slammer packets" UDP dest-port eq 1434 IP length eq 404 UDP payload-start offset 196 size 4 eq 0x4011010
class-map type access-control match-all
gnutella
IP
Nimda
TCP
Stack
Payload
match start TCP payload-start offset 0 size 32 regex "^GNUTELLA CONNECT
class-map type access-control match-any match match match match match match match
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cm-nimda1
regex regex regex regex regex regex regex "[\\/]csrss\.exe" "[\\/]httpodbc\.dll" "[\\/]sample\.exe" "[\\/]dnsservice\.exe" "[\\/]puta\.eml" "[\\/]puta\.scr" "[\\/]readme\.eml" 93
IP
TCP HTTP
Stack
start start start start start start start
l3-start l3-start l3-start l3-start l3-start l3-start l3-start
offset offset offset offset offset offset offset
40 40 40 40 40 40 40
size size size size size size size
32 32 32 32 32 32 32
Alternative to the FPM CLI Traffic Classification Definition Files

Traffic Classification Definition Files (TCDF) are published on CCO!
PISA(config)#load protocol bootdisk:ip.phdf PISA(config)#load protocol bootdisk:udp.phdf PISA(config)#load protocol bootdisk:tcp.phdf PISA(config)#loadBittorrent bootdisk:icmp.phdf protocol PISA(config)#load protocol bootdisk:ether.phdf PISA(config)#load classification bootdisk:fpmtemplate.tcdf PISA(config)#load classification bootdisk:bittorrent.tcdf IIS-Unicode PISA(config)#load classification bootdisk:skype.tcdf
IOS HTTP vuln. Skype

(www.cisco.com/cgi-bin/tablebuild.pl/fpm)
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Supervisor 32 PISA Day in The Life of a Packet

Supervisor 32 PISA Processing
Traffic is redirected to PISA when NBAR or FPM is configured on an interface PISA acts as bump in the wire service
PISA
NBAR/FPM Configured on G1/1 ingress
G1/1
G1/8 NBAR/FPM Configured on G1/9 egress
G1/2 G1/3 No PISA accelerated feature configured
G1/9 G1/10
Traffic Flow through PISA Traffic Flow bypassing PISA

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Supervisor 32 PISA Day in The Life of a Packet

Stateful versus Stateless Feature
NBAR is a stateful feature whereas FPM is a stateless feature NBAR needs to classify traffic not just in the direction where it is configured, but also in the reverse direction FPM is stateless and only needs to see traffic in the direction where it is configured High Availability design considerations:
Avoid asymmetrical traffic flows for NBAR SSO compatible: configuration synchronized. Flow state and statistics are not synchronized.
PISA
NBAR configured on G1/1 ingress
G1/1
G1/8
G1/2 FPM configured on G1/2 ingress
G1/9
L3 IPv4 unicast traffic Traffic is redirected to PISA when NBAR/FPM are configured Non L3 IPv4 unicast traffic bypass PISA
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Network Management TECUCT-1001, US Session BRKVVT-2005
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Cisco Unified Provisioning Manager

Single view of a subscriber and their services Simplified management of subscribers, services, and Cisco Unified resources
Day 1 template deployments Day 2 delegate MACDs
Single provisioning interface to Cisco Unified systems Policy-based, business oriented With the speed and accuracy of automation
CUCM, CME Unity, CUE, UC
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Cisco Unified Provisioning Manager

Request for Service
Provisioning Unified: ONE interface Simplified: business- p rocess & user o - riented Rapid: <1 minute for activation Accurate: Reduce manual & duplicate entry errors User friendly: template b - ased interface & business abstraction
Service Active
Prior to CUPM Manual System Provisioning UCME CUE Automated Activation UC Manager Clusters
Unity Voicemail Systems
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Cisco Unified Service Statistics Manager

Leverages wealth of data in Operations Manager and Service Monitor Enables partitioning of Unified Communications statistics data
CPU Utilization %
System CPU Utilization & Memory Used for CUCM SJX-1 on May 15, 2006
CPU Utilization % 6.00% 5.00% 4.00% 3.00% 2.00% 1.00% 0.00% 12:00 AM 5:00 AM 10:00 AM 3:00 PM8:00 PM Memory in Use (MB) 255 250 245 240 235 230 225 220 215 210 205
Based on network, service, business and user criteria System-wide, across multiple UC Manager clusters, Unity voicemail systems, UC Manager Express and gateways
Enables SLA creation, measurement and verification based on Unified Communications metrics Generates out-of-the-box reports
Specific focus on Executive, Operations and Capacity Planning roles
120% 100% 80% 60% 40% 20% 0%
Trunk Utilization Report for Trunk TRK-SJC-311 for Gateway GW-SJC-012 for 2006
Trunk Utilization % Poly. (Trunk Utilization %)

Trend line & extrapolation using Regression Analysis to predict future problems and fix proactively (upgrade Trunk TRK-SJC-311)
Enables export of data and reports to external applications

Ja n Fe ua br ry ua M ry ar ch Ap ri M l ay Ju ne Ju Se Aug ly pt us em t O be c r No to b ve er De m ce be m r be r
Provides ability to customize reports based on user needs
Memory in Use (MB)
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Service Statistics Manager

Advanced Performance Reporting
Report generation and display via Web-based user interface User-friendly interface featuring customizable dashboard at login Users can specify the reports to be shown on their dashboard and also specify layout (using a flexible Post-It paradigm) Intuitive navigation and drill-down reporting, from aggregate to fine levels of detail Flexible, customized report templates, ad-hoc reporting, in addition to default reports Automatic invocation & scheduling
e.g. Generate the report X every Friday at 5 PM, and e-mail the PDF report (or a web link to it) to the following list of people, but only if the following criteria match (utilization threshold exceeded, etc.)
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Security US session BRKVVT-2002
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Security
Breakout session: IP Communications Security: Threats and Mitigation Techniques New phone detection scheme
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Phone CDP Security

Switches can now check more than CDP to allow a device in the voice VLAN Can check for CDP and Line Power Can check for CDP, Line Power and Full Duplex 3560 and 3750 version 12.2(36) SE
IOS(config-if)# switchport voice detect cisco-phone

Check Line Power and CDP
IOS(config-if)#switchport voice detect cisco-phone full-duplex

Check Line Power, CDP and only Full Duplex
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Recommended Reading
Continue your Networkers at Cisco Live learning experience with further reading from Cisco Press Check the Recommended Reading flyer for suggested books A few suggestions:
Cisco CallManager Fundamentals, Second edition Troubleshooting Cisco IP Telephony
Available Onsite at the Cisco Company Store

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Complete Your Online Session Evaluation

Win fabulous prizes; Give us your feedback Receive ten Passport Points for each session evaluation you complete Go to the Internet stations located throughout the Convention Center to complete your session evaluation Drawings will be held in the World of Solutions
Tuesday, June 20 at 12:15 p.m. Wednesday, June 21 at 12:15 p.m. Thursday, June 22 at 12:15 p.m. and 2:00 p.m.
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New Developments in Unified Communications Design

2008 Cisco Systems, Inc. All rights reserved.

Session Scope and Objectives

2008 Cisco Systems, Inc. All rights reserved.

TCL/VXML Highlights from US session BRKVVT-2003

2008 Cisco Systems, Inc. All rights reserved.

Custom Applications Overview

Build custom applications to solve business problems or raise worker productivity

Tcl Application Architecture

H.323/SIP Joe Blow

dial-peer voice 2 pots direct-inward-dial application modify_name

dial-peer voice 1 voip destination-pattern 720555xxxx session-target ipv4:10.0.0.1

2008 Cisco Systems, Inc. All rights reserved.

Introducing Tcl Capabilities

Ability to manipulate CALL INFORMATION

Ability to manipulate call before CONNECT

CALL STATE control

VoiceXML Application Architecture

Answers call, fetches VoiceXML docs at URLs

2008 Cisco Systems, Inc. All rights reserved.

Introducing VoiceXML Capabilities

Ability to run on TDM or IP leg of call

VoiceXML browser has full VoiceXML 2.0 Compliance

Extensive IVR capabilities: Full control of user dialog flow

Recording of voice phrases or segments

Simple call control (e.g. call transfer and disconnect

Customized call routing

Tcl and VoiceXML Applications

Hybrid Tcl and VoiceXML Applications Leverage Best Capabilities of Both

History and Standards

Script Development and Support

Cisco Developer Support

Cisco TAC support

Sample Tcl/VoiceXML Scripts on Cisco.com

Tcl Script for CME/SRST

Tcl Programmer Guide sample scripts: Appendix A

Embedded Event manager (EEM) Tcl Scripts

Cisco Developer Support Tcl and VoiceXML Sample Scripts

2008 Cisco Systems, Inc. All rights reserved.

Voice at the Edge, or Core

Please enter your account number

Cisco Voice Gateway

Please enter your account number

Scaling, performance and cost benefits

Small-Medium Business: Basic ACD

Agent Group 1 Agent Group 2

2008 Cisco Systems, Inc. All rights reserved.

Enterprise: FXO Hookflash Transfer

Small Branch Office

Enterprise: Do-Not-Call List

CUCM Employee initiates outbound calls

2008 Cisco Systems, Inc. All rights reserved.

Enterprise: Distributed Contact Center

Call survivability at PSTN Gateway provided by Tcl script

Service Provider: Prepaid Calling Card

Gateways Highlights from US session BRKVVT2010

2008 Cisco Systems, Inc. All rights reserved.

le odu M SP D 2-64 ceable DM -PV Repla X AS5 Field

Medium Complexity High Complexity

H.320 ISDN Video Connectivity

Integrated Voice + Data Access

North-America: Serial Data