BRKCCT-2020_c1_cug

BRKCCT-2020

Planning and Designing a Cisco Unified Customer Voice Portal Deployment

Vipin Palawat

Chris Klardie

© 2011 Cisco and/or its affiliates. All rights reserved. Cisco PublicBRKCCT-2020 2

Session Overview

Fundamentals of planning and designing a Cisco Unified Customer Voice Portal (CVP) deployment

Details of various functional deployment models with their call flows, design details and case studies

This is an intermediate to advanced level session

Some familiarity with the Contact Center Enterprise products is required

Product Updates for CVP 8.5, FCS December 2010


Agenda

Platform and Virtualization

OAMP in CVP Deployments

CVP 8.0/8.5 Features, CUBE Support and Sizing

SIP Design, Branch Office Design

Architecture & Overview

Planning and Design

Serviceability

CVP Overview

CVP Components

Functional Models

Serviceability


About The Presenters

Vipin PalawatTechnical LeaderCVP ArchitectCustomer Contact Business Unit

Chris Klardie

ManagerEngineering and ArchitectureCustomer Contact Business Unit

CVP Overview



CVP – Main Features

Service Creation Queuing


CVP – Service Creation

Call Studio

Easy to use, yet flexible

Vibrant user community

<vxml version="2.0”>

<form id="form-record">

<record name="greeting" beep="true“>

<prompt>

Please record your greeting.

</prompt>


CVP – Queuing

SIP Call Control

Queuing at the edge saves money

Cisco network is the platform

P$TN


CVP Overview – Big Picture

Customers Agents

Four Pillars of CVP Solution


Service Creation Environment

Script Using ICM Script Editor

DTMF-only applications

Queue

Transfers to legacy IVRs/ACDs or CUCMcall center agents

Script Using Unified Call Studio

Dynamic VoiceXML self-service applications

Rapid Application Development

Eclipse-based

Connectivity to backend Web services

CVP Components“CVP requires a solution level mindset”



V 123

4

VXML

1 - New call from TDM GW to CVP2 - New call to UCCE from CVP3 - Play “Hello World” prompt4 - CVP sends call to VXML Gateway

Caller hears HW. Subsequent IVR5 - Agent is now available6 - CVP sends call to an agent

Caller is disconnected from IVRCaller is connected to an agent

5

6

Caller – Agent Conversation

IVR

CVP Components High Level


1

RTP

IP

JTAPI

CUIC

SIP

SS

Media Server

(Media Files)

IVR

SS

VXML Server

I

C

M

SS3

54

6

V

VXML

VXML

PG

HTTP

SIP Proxy can be optionally inserted at 1, 4, 6

2

1 - New call from TDM GW to CVP2 - New call to UCCE from CVP3 - Play “Hello World” prompt4 - CVP sends call to VXML Gateway

Caller hears IVR5 - Agent is now available6 - CVP sends call to an agent

Caller – Agent Conversation

ASR/TTS

RPT

CVP Component Details

Functional Models



Functional Models

CVP Standalone (optionally with ICM Lookup)

Automated Self Service IVR, No queuing, Limited call control, No formal agent

Call director

IP-enabled or TDM ACDs, TDM migration to IP, On-Net routing & transfer. Save $$

VRU-only

IVR and queuing. Call control via SS7.

Comprehensive

Pure IP-based contact center, IVR, call control, queuing

Basic service video

Audio-only IVR, call control, queuing, video agent


CVP Standalone

Caller (TDM or IP)

CVP VoiceXML Server

Voice Gateway /

VXML Browser

1. Incoming Call Leg

(SIP or TDM)

2. HTTP / VoiceXML

Documents

CVP Studio

SIP

HTTP


CVP Standalone

Terminates calls independently of UC Manager and ICM

Remotely distributed call delivery and IVR served by centralized applications avoids media across WAN

For advanced speech applications allowing customization and element reusability

Back-end integration flexibility (Database or web services)

Use with TDM or SIP

Basic Transfers (no queuing)


CVP Standalone with ICM Lookup

Caller (TDM or IP)

CVP VoiceXML Server

Voice Gateway /

VXML Browser


(SIP or TDM)

2. HTTP / VoiceXML

Documents3. NEW CALL Route

Request

SIP

GED-125

HTTP

PG


CVP Standalone with ICM Lookup

Typical application is to query the ICM only when the caller opts out to an agent

ICM Lookup can return a Translation Route label which CVP standalone could SIP REFER back into a CVP Comprehensive farm

CVP Standalone with ICM Lookup cannot process an *incoming* Translation Route label


CVP Comprehensive

PSTN

CVP Call Server

/VXML Server


(SIP or TDM)

5. SIP

INVITE

2. SIP INVITE

6. HTTP / VoiceXML

Documents

3. GED-125 New

Call

4. Connect to VRU

VXML Gateway

Ingress Gateway

CUCM

8. SIP

INVITE

7. Connect to

Agent

SIP

GED-125

RTP

INAP

HTTP

SS7


CVP Comprehensive

IVR, call control, queuing

ICM keeps track of agents

VXML Session: VXML-GW to Call Server (IVR Service) or VXML Server

Components:

Call Server

VXML Server (optional)

VXML Gateway

SIP Proxy (optional)

ICM and VRU PG

CUCM or TDM ACD

Reporting Server (optional)

ASR/TTS (optional)

CSS/ACE (optional)


CVP Basic Video


CVP Basic Video

Audio-only IVR, call control, queuing, video agent

Same call flow as comprehensive SIP

Supports both audio and video calls

Video supported with SIP only

Video Endpoints:

Cisco Unified Video Advantage

TelePresence™

7985G

Cisco H324M 3G Gateway

Platform and Virtualization

Planning and Design


HW Platform and OS Support

Call Server, VXML Server, Reporting Server must use MCS-7845 or higher server

Windows 2003 Standard supported for all components

VXML Server

Tomcat or WebSphere*

Operations Console

MCS-7825 or higher server

Unified Call Studio

Windows XP SP2/Windows Vista. Windows OS compatible HW

Windows 7 in Call Studio 8.5

* Several caveats


Hardware Components in CVP 8.5

UCS B-Series Blade Server (B200M1)

CVP Server Virtualization – UCS with VMware ESXi 4.0

Cisco Unified SIP Proxy (CUSP)

ISR based Proxy Server

Application Content Engine (ACE)

Next generation content engine used for load balancing

ISR G2 (2900 & 3900 Series)

Greater Performance

ISR G2 & 15.0(1)M - supported with CVP 8.0(1) as well

15.0(1)M IOS release required for ISR G2

UCS C-Series Rack Mount Server (C210M2)


Virtual CVP on UCS Solution Highlights

CVP in a Virtual (VMware) Environment

Supported with VMware ESXi 4.0

SIP deployments only

UCS hardware must be used

Specific VM configurations/templates

+

+

+


Virtual Machine Planning/Designing

No oversubscription of RAM or CPU!

Mix of Cisco UC applications on same blade – OK

Mixed environments of physical and virtual servers

Use published OVA templates (cores, memory and disk)

HA: Distribute Application nodes across UCS blades, chassis and sites

3rd party applications (Email, SQL, Web Server etc.) on the same blade with Cisco UC apps - not supported except Domain Controller and Media Server

3rd part app

servers


Virtualized CVP Design – Basic Rules

SRND application-layer guidelines for UCS are *not* same as when on MCS

Determine quantity or role of CVP nodes. Re-size for VM.

Private network requirement for VRU PG

Mixed clusters of bare-metal HP or IBM and UCS are supported

Subject to ―common sense‖ rules – e.g. don’t make Primary CVP Call Server less powerful than Secondary CVP Call Server or vice-versa

CVP OAMP

Planning and Design


CVP Operations Console

V

Call Server

VXML ServerRPT Server IOS Device UCM Device ICM Device CUIC

Device

CUP

Device

OAMP manages different device types of the solution


CVP Operations Console

Operations Console is used to provision and administer the CVP Solution. Ops Console is required to deploy CVP 4.0 and later releases

Monitor each CVP device and display statistics about each one

Start, shutdown or gracefully shutdown any of the unified CVP devices

CVP devices can only be managed by one OAMP Server


CVP Operations Console System Configuration

CVP OAMP now provides administration of various system-level configurations.

A system-level configuration allows an administrator to manage configurations that can be deployed to multiple devices in a single operation.

This differs from device configuration as device configuration allows

for deployment of a single configuration to a single device.


CVP Operations Console Network Map

The CVP Operations Console Server now generates and distributes a network map to all configured CVP devices as well as other, third-party managed devices

The network map contains information which allows serviceability tools (such as the Analysis Manager and System CLI) to access individual devices from the network map

Through generation of the network map, the CVP Operations Console Server serves as a seed device

With proper authentication, the Analysis Manager and System CLI tools can query the CVP Operations Console Server to provide information about configured devices in the Operations Console Server. An example of information provided include device IP addresses, hostnames, and credential configuration.


CVP OAMP Device Management

General

Device Versioning: This field will display the device version of the target CVP device

Device Specific

Unified CVP Call Server

Advanced Configuration - Unified CVP Call Server configuration in the ICM, SIP, and IVR tabs have been rearranged to only display common configuration fields.

Can now add a new CallServer with just IP/Hostname and selection of service (SIP, ICM, IVR)

Unified CM, ICM, SIP Proxy, IC

Can provide serviceability credentials


OAMP Bulk Administration

Two click operation to transfer all default gateway files to all provisioned gateways.

Bulk administration file transfers will now occur in the background so Operations Console Server administrators can manage other tasks while a transfer is in progress.


CVP OAMP Bulk Administration

CVP Server licenses can be combined into a single license file and can be bulk uploaded to all servers in a simple point, click, Transfer operation.

CVP servers can intelligently determine the proper license to use.

Point, click, Transfer and update multiple applications to all or selected VXML Servers by providing a single application zip file from CVP Studio.


CVP OAMP IOS Templates

Pre-installed IOS Configuration Templates

Allows custom IOS Configuration Templates

Automatically pulls the information from configured devices

Requires template variables to follow a naming convention

Allows in-context editing of template before deployment

Pushes the configuration to multiple IOS devices in a single operation.

Maintains transactional integrity on a single IOS device

If there are any failures, it will automatically rollback on that specific device.

Rollback the last configuration on a single or multiple devices.


IOS Template Example for CVP


CVP OAMP System Dialed Number Pattern

In previous releases of CVP, one can configure SIP Dialed Number (DN) patterns in the SIP tab of the Call Server configuration.

Unfortunately, the usability for configuration of the DN patterns were somewhat limited. In addition, there wasn't a way to configure system-level DN patterns which would apply to the entire CVP deployment

To address these limitations, CVP 8.5 provides functionality that enables one to apply DN patterns to the entire CVP deployment via a system-level configuration


CVP System Dialed Number Pattern (Continued)

The following lists some of the new functionality:

Provides system-level deployment of DN patterns

Provides ability to configure large number of DN patterns

Provides ability to sort and filter DN patterns

DN patterns may be specified at a system level or at a Call Server device level (as in previous releases)

The choice of whether to use DN configured at system level or Call Server device level can be made in the SIP tab of Call Server device

UI option of nested tree or a simple list view


Media Server

The CVP OAMP Media Server device was enhanced to populate the media server device list to all Call Servers.

The Media Server device list can be updated to all the CallServers by pressing the new "Deploy" button from the OAMP Media Server device list.

A default media server device may be specified in OAMP. If specified in OAMP, micro-applications will use that default media server if the ECC variable for media server is not defined in the UCCE ICM script. The default media server is specified in the OAMP Media Server list page.

Supports primary and backup media server via hostnames

Supports ACE if there are multiple media servers

Supports FTP on media server for agent greetings


Media Server (Continued)

Due to default media server, ICM scripting is now simplified

Media server is defined by ECC variable

Default media server is configured in CVP OAMP ( Allowed in Release 8.5(1) )


Media Server (Continued)

Traditional VXML Server application in ICM script

Simplified ( Allowed in Release 8.5(1) )

CVP 8.0 Features

Planning and Design


DS0 Trunk Information

CVP 8.0 release adds the capability of passing the PSTN gateway trunk and DS0 information (on which a call arrived) to ICM

Supported with SIP calls only

Can be used to identify DS0 for a call from ICM reports

Can be used in ICM Routing Script as well

The PSTN trunk group data will come from the PSTN Gateway in SIP

INVITE message as given below:

Via: SIP/2.0/UDP 192.168.1.79:5060;x-route-tag="*tgrp:2811-b-000"*;x-

ds0num="ISDN 0/0/0:15 0/0/0:DS1 1:DS0";branch


Trunk Utilization and Dynamic Routing

This feature allows the gateways to push status of memory, DS0, DSP and CPU to CVP

Utilization of resources are sent

On periodic interval

When changes to high or low level watermark

When requested by CVP

Can be used to do trunk utilization reporting at ICM

Can be used in ICM Routing Script as well

Status used in CVP SIP Server group to mark a SIP element as UP/DOWN instantaneously

CVP reporting reports on DSP, CPU, Mem in addition to DS0


Post Call Survey (PCS) for SIP

Post Call Survey with SIP

Already available with H323

After the agent hangs up, the caller is then transferred to a DNIS that will prompt the caller with a Post Call Survey

Caller is sent to a PCS DN based on the original dialed number

PCS call: another regular call from UCCE point of view

Call context information retrieved from the original customer call

OAMP System Dialed Number

Example: If the original called

DN was 2233*, the caller will be

sent to survey DN of 8000


Key Press Markup Language (KPML)

KPML – out of band SIP DTMF signaling method

The key press information is passed in the SIP signaling instead of the RTP stream

RFC2833 is widely deployed DTMF method with SIP CVP

Legacy TDM transfers (DTMF *8) not supported with KPML via CVP

Some endpoints do not support in-band RFC2833

7985 IP phone, and CTI ports in IPCC mobile agent support KPML

For these endpoints, when the destination behind the SIP Trunk is set with RFC2833, CUCM allocates an MTP resource to translate in-band RFC2833 DTMF packets to out of band KPML signaling messages

Don’t assume that KPML support eliminates all the need for an MTP


MMOH & SIP Header Passing to ICM

Multicast music-on-hold (MOH) Allows true queue at edge

Passing information in SIP headers to ICM

Users can specify SIP header values in outgoing SIP INVITES only

Only the initial SIP INVITE, not re-invites

No syntax checking while adding or modifying headers in OAMP

SIP header modification feature is a powerful tool which can tweak SIP headers as needed. Users should exercise caution


Courtesy Callback Overview

Reduce the time a caller spends on the phone waiting for an agent to answer.

Instead of waiting and listening to queue music, the caller has the option to request a callback at a chosen phone number, without losing his/her place in queue.

As the Estimated Wait Time (EWT) approaches zero, a call is initiated to the caller at the given phone number.

With a short wait in queue, the caller is transferred to an agent.

• Courtesy Callback (CCB) reduces the time callers have to wait on the phone.


Queue

Callers

The caller’s queue

position is preserved

and appears the

same to agents

answering the call.

Courtesy Callback Overview


Courtesy Callback

The scientific explanation


Courtesy Callback Decision Flow

Call Back?

CallerAvailable

?

Caller is put in queue

Caller providesphone # for

callback

caller waits in queue

EWT approaches 0

Callback initiated

Agent available

?

Caller’s queue position

preserved. Call disconnects.

Caller gets connected to agent

Callback terminates after 3 attempts

(default)

YY

Y

NN

N


Components Needed for CCB


Determining When to Call Back

CCB is designed to call back callers based on contact center dynamics such as:

• The number of calls already in queue

• The average handle time for calls

• The number of agents in ready and talking state

Based on this information, CCB continually computes the wait time in queue for each caller and initiates the callback when the wait time approaches the agent availability mark


Service Level Tuning

Service level is the average time that callers wait on the phone for an agent after callback. It is adjustable using configurable parameters.

Setting the appropriate service level is important so that callers are not called back too early or too late, as both scenarios are undesirable.

CVP 8.5 Features

Planning and Design


Agent Greeting

An Agent Greeting is a recorded announcement that plays when a caller and an agent are successfully connected for an inbound (caller to agent) call.

The call can be a queued or a non-queued call. The greeting is recorded as a wav file which is stored on the Unified CVP's media server.

Using a recorded greeting allows you to present a clear, well-paced, calm, and enthusiastic voice to every caller without having to repeat the same phrase over and over during the course of a day.

Using Agent Greeting also gives the agent time to review the screen pop and reduces the noise level in the call center. Both the agent and the caller hear the greeting.

Agent Greeting is turned on (is enabled) by default at agent login time.


Agent Greeting Use Case

Thank you

for calling

Super

Tablet…

Thank you

for calling

Super

Tablet…

Thank you

for calling

Super

Tablet…

Thank you

for calling

Super

Tablet…

Let’s get

that camera

fixed

Using a recorded greeting allows

you to present a clear, well-

paced, calm, and enthusiastic

voice to every caller without

having to repeat the same

phrase over and over during the

course of a day.


Agent Greeting

Step 1 - 5: Regular call delivered at agent

Step 6 - 8: Agent greeting triggered at UCCE as soon as agent answers the call and when UCCE determines that

this specific agent requires an agent greeting.

Step 6-8 is disconnected (by CVP) at the end of agent greeting playback. Caller and Agent continues the normal

conversation after agent greeting.

Step 6 invokes the JTAPI "AddMedia" API introduced in 8.5.1 release to trigger agent greeting call on phones

that support BIB.


Agent Greeting – How to Use It

An agent who is logged in to the CTI OS Agent Desktop and is in the Not Ready state can press a desktop icon to record a greeting.

Pressing this icon initiates a call to Unified CVP in which the agent is prompted to record, listen to, and save the greeting, similar to recording a voice-mail greeting.

When the agent confirms and saves the greeting, it is saved as a wav file with a file name that includes the agent ID and the Greeting Type (specified in the Unified CCE routing script that saves the greeting). For example, agent001_english_sales.wav.

Refer to Sizing tool for Agent Greeting sizing impact.


ICM Scripting for Agent Greeting

Once the agent has recorded a greeting, the administrator must add a

Set Variable node to the routing script that directs calls to that agent's

skill group

The Set Variable node identifies the Greeting Type to use.


Agent Greeting – Known Caveats

Agent Greeting is not supported in Outbound Dialer or Parent-Child deployment models.

Agent Greeting is only supported with CVP and using the SIP call control protocol.

Agent Greeting is only supported with phone models with BIB

UCM 8.5(1), CCE 8.5(1), CTIOS 8.5(1), and CVP 8.5(1) are required.

The CTIOS desktop has an "Enable/Disable" greeting button. But the CAD/CSD desktops do not have an "Enable/Disable" greeting button; customization is required to add a button to dial CVP for recording a greeting

Remote Agents (agents from home) - not supported in 8.5(1).

Mobile Agents - not supported in 8.5(1)

Finesse Desktops - Not supported in 8.5(1).

Whisper Announcement

Planning and Design


Whisper Announcement

Whisper Announcement is defined as the ability to play a pre-recorded announcement to an Agent right before the caller is connected.

The whisper announcement is played to the Agent only. e.g. ―Shipment Ground English" or "InkJet Cartridge Replacement". This announcement helps the Agent answer the call correctly.

The announcement is not heard by the Caller. The caller hears ringing while a whisper announcement is in progress.

Whisper announcement time is counted as ring time (not talk time).


Whisper Announcement CallFlow


Whisper Announcement ICM Scripting

The administrator needs to indicate, in a routing script, which whisper announcement to play.

A new "WhisperAnnouncement" call variable has been created for this purpose (note: not an ECC).

Use a Set Variable node to indicate the particular whisper announcement audio file that should be played to the agent.

Cisco Unified Border Element Support

Planning and Design


CVP and CUBE

CUBE acts as a demarcation point

Co-resident VXML Gateway and CUBE SBC is supported on ISR only with sizing changes

SIP-SIP translation only is supported

PSTN/3rd party SIP Trunks

CUBE is supported in flow-through mode only

survivability.tcl script and services are supported

Mixed codec deployments supported in ISR/5xxx GW only


ASR1000 as CUBE

ASR doesn’t support VXML. Separate VXMLGW required.

Stateful failover is only supported with UDP

Mid call codec re-negotiation is not yet supported

Only flat G.711 codec supported

PSTN GW features are not supported. Examples,

Legacy transfer connect

Trunk Group Utilization

DS0 information

Hookflash

TBCT


SIP REFER: CUBE Consumes

Caller (TDM or IP)

CVP Call ServerCUBE

4. SIP INVITE

xxxxxxx

3. SIP REFER to

xxxxxxx2. CONNECT to

RFxxxxxxx

1. Must first ―Send to

VRU‖ and then create

RFxxxxxxx label script

node

On CUBE:

voice service voip

no supplementary-service sip refer

Call must first be in connected state to perform REFER (hence the SendToVRU as the first

step in the ICM script). If not done, a 302 REDIRECT will occur instead of a REFER.

The destination that CUBE routes xxxxxxx to is defined in the CUBE dial-peer for xxxxxxx.

This will override whatever destination was chosen by CVP in the Refer-To header of the

REFER.

SIP

GED-125


SIP REFER: CUBE Pass-Through

Caller (TDM or IP)

CVP Call ServerCUBE

5. SIP INVITE

xxxxxxx 3. SIP REFER

to xxxxxxx2. CONNECT

to RFxxxxxxx

1. Must first “Send to

VRU” and then create

RFxxxxxxx label script

node

4. SIP REFER

to xxxxxxx

On CUBE:

voice service voip

supplementary-service sip refer

SIP

GED-125

Sizing

Planning and Design


CVP Component Sizing ( Non UCS/VM )

ComponentScalabilityper Server

Redundancy HW Platform

Call Server (SIP) 1200 ports, 15 cps N+1 to N*2 MCS-7845

Call Server (H.323)500 ports

7 cpsN+1 to N*2 MCS-7845

VXML Server 1200 ports, 15 cps N+1 to N*2 MCS-7845

Operations Console NA NA MCS-7825 or higher

Co-Res (SIP, VXML, Media Server)

1200 SIP ports + 1200 VXML ports

15 cps

N+1 to N*2 MCS-7845

Co-Res (H323, VXML, Media Server)

500 H323 ports + 500 VXML ports

N+1 to N*2 MCS-7845

VRU PG 9600 ports N*2 MCS-7845


Supported VM Templates for CVP

Component & Scale * vCPU vRAM (GB) vDisk (GB)

Call+VXML+Media (900 Calls, 10 cps) 4 4 1 x 80

Reporting (Med) (420 Msg/sec) 4 4 1 x 364

Reporting (Large) (420 Msg/sec) 4 4 1 x 510

OAMP Server 2 2 1 x18

* CPU referenced is 2.53GHz Xeon E5540, RAM 1066MHz


VXML Gateway SizingMaximum VXML Sessions: 12.4.15T

The numbers assume the only activities running on the GW are VXML with basic routing and IP connectivity. These figures apply to NTE 75% CPU, VAD off, Cisco IOS 12.4.15T5. The numbers represent performance with Call Studio generated scripts running on CVP VoiceXML Application Servers. VoiceXML 2.0 and MRCPv2 were tested. The 1861 requires 12.4.20T as a minimum release.

*NTE 80% CPU.

Voice Gateway Platform

Dedicated VXML GW Voice Gateway + VoiceXML

All calls VXML-Controlled; no PSTN interfaces present

All calls are PSTN calls and all calls are VXML-Controlled

DTMF ASR/TTS DTMF ASR/TTS

1861 5 4 4 2

2801 7 6 6 4

2811 30 24 25 20

2821 48 36 36 30

2851 60 56 56 48

3825 180 140 210 130

3845 200 155 230 145

AS5350XM* 240 192 240 160

AS5400XM* 240 192 240 160


VXML Gateway SizingMaximum VXML Sessions: 15.0.1M


Dedicated VXML GW Voice Gateway + VoiceXML

All calls VXML-Controlled; no PSTN interfaces present

All calls are PSTN calls and all calls are VXML-Controlled

DTMF ASR/TTS DTMF ASR/TTS

1861 5 3 4 2

2801 7 4 5 3

2811 30 20 23 15

2821 48 32 36 25

2851 60 40 45 30

3825 130 85 102 68

3845 160 105 125 83

5400XM 200 135 155 104

2901 12 8 9 6

2911 60 40 47 31

2921 90 60 71 48

2951 120 80 95 64

3925 240 160 190 127

3945 340 228 270 180

29xx and 39xx are ISR G2

15.0.1M, G.711, basic calls, Ethernet

egress, CPU NTE 75%, VAD is ON


CUBE Gateway SizingMaximum VXML Sessions: 15.0.1

15.0.1M, G.711, basic calls, Ethernet egress, CPU NTE 75%, VAD is ON


Dedicated CUBE GW CUBE + VoiceXML

Flow-Through with VAD OFF

Flow-Through with VAD ON

Flow-Around All SIP Calls & VXML controlled with Flow-Through & VAD ON

DTMF ASR/TTS

2801 55 75 200 5 3

2811 110 150 400 23 15

2821 200 300 600 36 25

2851 225 325 750 45 30

3825 400 500 750 102 68

3845 500 600 750 125 83

AS5000XM 600 850 3000 155 104

2901 100 130 400 9 6

2911 200 260 800 47 31

2921 400 520 1500 71 48

2951 600 780 2500 95 64

3925-SPE100 800 1000 3000 190 127

3945-SPE150 1000 1250 4500* 270 180

CVP 7/8


CUSP Sizing with CVP

SRE License Standard Mode

(record route on)

Requests per second

Lite mode (record route off)

Requests per second

FL-CUSP-2 2 5

FL-CUSP-10 10 25

FL-CUSP-30 30 75

FL-CUSP-100 100 450

FL-CUSP-200 200 750

CVP 8.5

NME License Standard Mode

(record route on)

Requests per second

Lite mode (record route off)

Requests per second

FL-CUSP-10 10 10

FL-CUSP-30 30 30

FL-CUSP-100 100 450


CUSP Sizing with CVP (Continued)

CVP 8.5

Performance based on 1000 routes / no normalization

For CUSP sizing purposes, assume each CVP Call uses on average 4 SIP CUSP requests:

GW to CVP Call Server

CVP Call Server to VXML gateway

CVP Call server to Ringtone gateway

CVP Call server to CUCM

SIP Design

Planning and Design


SIP Service

SIP is the preferred protocol for CVP

Only SIP (no H323) allowed for new deployments from CVP8.0

H323 to be totally deprecated in CVP9.0

SIP provides improved scalability and performance to CVP

SIP and H.323 can co-exist at the same time (valuable during H323->SIP migration)

SIP Proxy, DNS SRV or static routes replace Gatekeeper function

SIP packets tagged with proper QoS markings vs. ACL for H.323

CVP 8 supports RFC 2833 (in-band RTP NTE) as well as KPML(OOB SIP-Notify) for DTMF

MTP will be allocated if IP phones and SIP trunk on CUCM are not configured for, or do not support RFC2833


SIP Routing

SIP Routing is the method in which a DN is resolved to an IP

Also used for load-balancing and redundancy

Several options available:

SIP Proxy

DNS SRV

Local SRV

Static routes

Combination of above

SIP Proxy desirable for significant number of SIP user agents or for complex dial plan.

Static routes/local SRV desirable for small deployments with simple dial plan.


Different Types of SIP Routing in CVP 8.0

CVP 8.0 release introduces more features and methods for configuring the dial plan and SIP call routing.

SIP routing in CVP 8.5 solution includes

Location based routing

SigDigits

Send To Originator

Outbound Proxy

Local SRV

Dynamic Routing


Algorithm and Routing PriorityWhen a Label Is Received By CVP from UCCE

Location SiteId (if present) is inserted for Comprehensive callflows before selecting a destination

SigDigits will be prepended if defined before routing to a destination

If SendToOriginator is matched for label, the callers source IP/host is used ONLY if caller is a Cisco IOS gateway

If (use outbound proxy) is set, then use the host of the SIP proxy

If local static route is found, then use the destination as SIP URI

If local SRV and SIP server group is defined, then destination is determined dynamically based on SIP element status and priority/weight


SIP High Availability

SIP HA can be achieved in multiple ways

Redundant SIP Proxies

DNS SRV records

Local SRV records

Static routes

Combination of above

As of CVP8.0 TCP is preferred transport for SIP with CVP. Was UDP prior to CVP8.0

CVP running on VMware has issues with UDP


OAMP Server Groups replace local SRV manual file creation. srv.xml file still used under the covers.

CVP knows the status of the destination before sending SIP INVITE using the SIP OPTIONS heartbeat feature

SIP Server Group and Dynamic Routing

CVP 8.0


SIP Server Group


SIP Proxy Server

Centralized dial-plan

Redundancy and load-balancing

Two proxies supported for contact center:

CUSP (preferred): runs on ISR blade

CUPS: runs as a cluster on CUCM OS

CVP Call Server must use SRV to use redundant SIP Proxies

Record-route should be disabled on Proxy

Otherwise all signaling passes through the Proxy, Proxy failure causes call failure


Cisco Unified SIP Proxy (CUSP)

Network Module (NME) in ISR Routers

3800, 2951, 3900, 3900E routers only

Service Module (SRE) in ISR Routers

2900(any), 3900, 3900E

Prior to 8.5 CUSP requires dedicated chassis and can not co-reside with VXML/TDM GW. 8.5 and after allows co-residency.

Redundancy: Two ISR GWs geographically separated

Double Capacity with the above by using 2 CUSP/ISR

CVP 8.5

CUSP

VVVV

Ingress GW

SIPSIP

SIPSIP

ISR

CUCM

Branch Design

Planning and Design


Branch Office Deployment


Branch DesignDesign for Queuing at the Edge

If call comes into remote site, it is desirable for the call to be queued at that site

CVP normally load-balances among all VXML gateways—not optimal for branch designs

―Send To Originator‖ sends the call back to same GW for VXML in SIP

―settransferlabel‖ causes CVP to always transfer back to originating gateway in H.323 case


Call Admission ControlWhy Is Call Admission Control (CAC) Needed?

IP WAN

Router/

Gateway

Call

Manager

PSTN

Circuit-Switched Networks

Packet-Switched Networks

PBX

Physical

Trunks

STOP

IP WAN

Link

IP WAN links LLQ Is

Provisioned

for 2 VoIP Calls (Equivalent

to 2 “Virtual” Trunks)

Third Call

Rejected

No Physical Limitation

on IP Links.

3rd Call Can Go Through,

but Voice Quality of All

Calls Degrades.

Call Admission

Control Blocks 3rd Call


Branch DesignCall Admission Control: The Problem

From the perspective of CM, the gateways are ―behind‖ the CVP server

CM treats every call as if they came from the same device (SIP Proxy/CVP/GK) based on source IP

Affects: locations CAC

PSTN

IP WAN

Headquarters

Branch A

Branch B

Plenty of BW


Branch DesignQueue at Edge : CAC Not an Issue Till This Point

Call received from PSTN, sent to CVP

CVP receives label for VRU, CVP does GK /SIP Proxylookup to find destination gateway

One VRU label is used, destination VXML-GW could be any

For SIP ―Send To Originator‖ causes CVP to transfer back to originating gateway

With these commands, CAC is not an issue for VXML leg

PSTN

IP WAN

Headquarters

Branch A

Branch B

1

2

3


Location Awareness and Location Based Routing

Without Location Awareness, You Will Have

Bandwidth miscalculations in CAC with IP originated callers, as well as with any post transfers from agents

Inability to deterministically select a local VXML GW for VRU treatment at the branch office during warm transfers from an agent due to no correlation between the two calls at consult


CVP gets the location information from CUCM, associate it to a particular branch gateway and assign a siteID.

CVP has information about all the gateway locations now

For SIP: CVP passes the location information in the ―Call-Info Header‖

Location Based CAC


CVP gets the location information from CUCM, associate it to a particular branch gateway and assign a siteID.

CVP has information about all the gateway locations now

CVP can now select the correct and closest VXML gateway (near to the caller) to queue the call

Saves WAN bandwidth

Proper SIP proxy configuration is required to route label based on the site-id information

Valid for IP originated calls as well

Location Based CAC SolutionAnother Advantage – Deterministic Selection of VXML Gateway

Serviceability


Diagnostic Portal API Overview

Diagnostic portal API is designed for UC products to publish diagnostic data

When the WSM receives a request, it constructs a XML response

WSM users are configured on OAMP (default wsmadmin created during CVP Install)

Analysis

Manager

System

CLI

CVP

WSM

https

https

Call Server

VXML Server

Ops Console

….


Analysis Manager or System CLI

Analysis Manager

Built inside UCM RTMT tool

GUI based

Supports Analyze Call Path

Unified System CLI

Installed as part of CVP and UCCE

No additional server required but separate server supported

Command line based and supports batch file execution

Support multi-instance ICM deployments


Analysis Manager or System CLI


System CLI – Simple Yet Powerful

System CLI is automatically installed on all CVP servers as part of the infrastructure, there is no additional installation required on any CVP server.

Every CVP server is also aware of at least one seed device (CVP Operations Console server), the entire solution topology is automatically retrieved from the Operations Console on any CVP box for System mode.

There is no additional configuration needed for system mode.

System CLI uses a consistent command across multiple products and servers.

System CLI can be executed as a Windows scheduled job.


Unified System CLI – Quick Start

Launch the CLI on any CVP server

Login with username ―wsmadmin‖ and password as OAMP Administrator password.


Unified System CLISample of a Local Collection

To collect all of the diagnostic information from your CVP server, type the "show tech-support" command in System CLI


Unified System CLI

By default ―show tech-support‖ collects the traces for the last 24 hours. For

example, if you want to pull traces for the last 3 days, use the following

command:

admin:show tech-support reltime 3 days

To collect all of the diagnostic information from all devices [CVP, IOS, UCM,

ICM, CUP, etc ] defined in CVP OAMP, type the "system show tech-support"

command in System CLI as shown below:

admin:system

Initializing system mode ...

Retrieving device list. This process may take a few

minutes to complete.

......................................

admin(system):show tech-support


System CLI Modes of Operation

System CLI can operate in two modes: local mode and system mode.

In the local mode, the System CLI only interacts with a single device. For example, "show version" command shows only the version for a single device. In the system mode, the System CLI interacts with all of the devices in the device list.

All of the commands available in local mode for a single device are available in system mode. The command syntax remains the same in system mode.

For example, in system mode, the "show version" command will show the version information from all devices in the device list.

In system mode, there is an additional option to limit the system command option to certain device group, device type, or list of servers.


Unified System CLISystem Mode Sample Collection

Enter System Mode

Execute a command


Unified System CLI – System Debugs

System CLI provides a consistent debug level across the solution

The definition of debug levels are given below for reference:

level 0 --- Default ( normal production environment )

level 1 --- Low performance impact

level 2 --- Medium performance impact

level 3 --- High performance impact

Level 4 and 5 are only for lab or functional use only

Level 99 (custom) is only meant for special situations

Example: Turn ―Detailed‖ traces on UCM

Command: ―system debug level 3 devicetype ucm component "ucm:Cisco CallManager"


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BRKCCT-2020_c1_cug

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