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© 2012 Cisco and/or its affiliates. All rights reserved. Presentation_ID Cisco Public Planning and Designing a Cisco Unified Customer Voice Portal Deployment BRKCCT-2020

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© 2012 Cisco and/or its affiliates. All rights reserved. Presentation_ID Cisco Public

Planning and Designing a Cisco Unified Customer Voice Portal Deployment BRKCCT-2020

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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 a migration case study

This is an intermediate to advanced level session

Some familiarity with the Contact Center Enterprise products is required

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Agenda Platform and Virtualization

Sizing

Bandwidth, QoS and Latency

Geographic Models

Edge Queuing Techniques

Architecture & Overview Planning and Design CVP Overview

CVP Components

Functional Models

Ingress and VXML Gateways

SIP including CUPS and CUSP

CVP Call Servers

Load Balancers (ACE and CSS)

Survivability

High Availability Designs

Case Study

Better Late than Never

Trust but Verify…

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About The Presenters

Rue Green Technical Leader Customer Collaboration Service Line Cisco Systems Advanced Services

Rahul Maniktala Consulting Systems Engineer Connected Architectures Partner Organization Cisco Systems

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CVP Overview

Architecture & Overview

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CVP – Main Features

Service Creation Queuing

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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>

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CVP – Queuing

SIP Call Control ‒ Queuing at the edge saves money

‒ Cisco network is the platform

P$TN

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CVP Overview – Big Picture

Customers Agents Four Pillars of CVP Solution

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CVP Components

“CVP requires a solution level mindset”

Architecture & Overview

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V 1 2 3

4

VXML

1 - New call from TDM GW to CVP 2 - New call to UCCE from CVP 3 - Play “Hello World” prompt 4 - CVP sends call to VXML Gateway

Caller hears HW. Subsequent IVR 5 - Agent is now available 6 - CVP sends call to an agent Caller is disconnected from IVR Caller is connected to an agent

5

6

Caller – Agent Conversation

IVR

CVP Components High Level

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1

RTP IP

JTAPI

CUIC

SIP SS

Media Server (Media Files)

IVR SS

VXML Server

I C M

SS

3 5

4

6

V

VXML

VXML

PG

HTTP

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

2

1 - New call from TDM GW to CVP 2 - New call to UCCE from CVP 3 - Play “Hello World” prompt 4 - CVP sends call to VXML Gateway

Caller hears IVR 5 - Agent is now available 6 - CVP sends call to an agent

Caller – Agent Conversation

ASR/TTS

RPT

CVP Component Details

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Functional Models

Architecture & Overview

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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

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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

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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)

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CVP Standalone with ICM Lookup

Caller (TDM or IP)

CVP VoiceXML Server

Voice Gateway / VXML Browser

1. Incoming Call Leg (SIP or TDM)

2. HTTP / VoiceXML Documents

3. NEW CALL Route Request

SIP

GED-125

HTTP

PG

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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

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CVP Comprehensive

PSTN

CVP Call Server /VXML Server

1. Incoming Call Leg (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

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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)

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CVP Basic Video

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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

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Platform and Virtualization

Planning and Design

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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

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Hardware Components in CVP 8.5

UCS B-Series Blade Server (B200M2) ‒ 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)

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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

+ +

+

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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

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Supported VM Templates for CVP

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

** Router 8K agents 4 4 1 x 50

** Logger 8K agents 4 4 1 x100

** Agent PG 2000 agents 2 4 1 x 50

** VRU PG 9600 ports 2 4 1 x 50

Supported VM Templates for UCCE - Example

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 ** Requires 2 vNICs. All others only require single vNIC.

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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

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Sizing

Planning and Design

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CVP Component Sizing ( Non UCS/VM )

Component Scalability per 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 cps N+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

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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

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VXML Gateway Sizing Maximum 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

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VXML Gateway Sizing Maximum VXML Sessions: 15.0.1M

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 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

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

Voice Gateway Platform

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

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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

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CUSP Sizing with CVP (Continued)

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

CVP 8.5

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Bandwidth, QoS and Latency

Planning and Design

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Call Signaling and Voice Bearer Traffic

Assumes typical H.323 call flow uses about 7000 bytes per call ‒ Every call into a branch gateway requires 6000 bytes plus 1000 bytes for each

transferred call to an agent.

‒ (7,000 bytes / call) * (8 bits/byte) = 56,000 bits per call

‒ (cps) * (56,000 bits / call) = Average kbps per branch

Assumes typical SIP call flow uses about 17,000 bytes per call ‒ (17,000 bytes / call) * (8 bits/byte) = 136,000 bits per call

‒ (cps) * (136,000 bits / call) = Average kbps per branch

‒ Remember, there are 4 SIP call legs per average UCCE call

For Voice Bearer Traffic calculations depend on the codec used – G.711 or G.729

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VXML Traffic 1 IVR prompt = 1 VXML document

Average VXML document = 7,000 bytes

7,000 bytes * 8 bits = 56,000 bits per prompt

(cps) * (56,000 bits / prompt) * (# of prompts / call) = kbps/branch

VXML Document Type VXML Document Size (Approximate)

Root document (one required at beginning of call)

19,000 bytes

Subdialog_start (at least one per call at beginning of call)

700 bytes

Query gateway for Call-ID and GUID (one required per call)

1,300 bytes

Menu (Increases in size with number of menu choices) 1,000 bytes + 2,000 bytes per menu choice

Play announcement (simple .wav file) 1,100 bytes

Cleanup (one required at end of call) 4,000 bytes

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Media File Retrieval Traffic

Assumes HTTP Traffic

(# prompts) * (size (bytes)/prompt) * (8 bits/byte) = Total bits

(Total bits) / (cache refresh interval (secs)) = Average kbps per branch

Example: 25 prompts, average size 50kB each, refresh int. 15 mins.

(25 prompts) * (50,000 bytes/prompt) * (8 bits/byte) = 10,000,000 bits

(10,000,000 bits) / (900 secs) = 11.1 Average kbps per branch

Not caching prompts at the VXML gateway causes significant Cisco IOS performance degradation as much as 35% to 40%, directly impacted the overall bandwidth budget for the solution.

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*Equations Used to Find These Values Can Be Found in the SRND

Sample Bandwidth Usage

CPS BHCA Agents Queue Time VXML Ports H.323 SIP VXML

.03 100 6 45 sec 9 2 kbps 4 kbps 40 kbps

.14 500 21 27 sec 18 8 kbps 19 kbps 106 kbps

.28 1000 38 24 sec 27 16 kbps 38 kbps 168 kbps

1.39 5000 172 17 sec 84 78 kbps 189 kbps 600 kbps

Based on the following assumptions:

IVR prompt and collect = 30 seconds, 1 Prompt every 6 seconds

Agent talk time = 120 seconds

CPS above assumes even distribution of BHCA, worst case CPS should be used

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CVP QoS Class Model

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CVP Port Usage and QoS Setting Component Port Queue PHB DSCP Maximum Latency

(Round Trip)

Media Server TCP 80 CVP-Data AF11 10 200 ms

Unified CVP Call Server TCP 1720 (h.323)

TCP/UDP 5060 (SIP) Call Signaling CS3 24 200 ms

Unified CVP IVR Service TCP 8000 (HTTP)

TCP 8443 (HTTPS) CVP-Data AF11 10 200 ms

Unified CVP VXML Server TCP 7000 (HTTP)

TCP 7443 (HTTPS) CVP-Data AF11 10 200 ms

Ingress Gateway TCP 1720 (H.323)

TCP/UDP 5060 (SIP) Call Signaling CS3 24 200 ms

VoiceVXML Gateway TCP 1720 (H.323)

TCP/UDP 5060 (SIP) Call Signaling CS3 24 200 ms

H.323 Gatekeeper UDP 1719 Call Signaling CS3 24 200 ms

SIP Proxy Server TCP/UDP 5060 Call Signaling CS3 24 200 ms

MRCP TCP 554 Call Signaling CS3 24 200 ms

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Network Latency

Latency < 200 ms round trip is recommended

Solution will tolerate latency as high as 400ms round trip

Since HTTP is primary protocol used to fetch VXML instructions and media files, HTTP should treated with higher priority than normal HTTP

HTTP traffic should be treated as CVP call signaling traffic

Lets take a look closer look at a WAN with ~250 ms round-trip delay….

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Network Latency Example

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VXML Call Prompt Setup Times – ICM Micro Apps

WAN RTT Latency SIP Call Setup Micro App Download (Initial VoiceXML) Time Until Prompt Plays

50 ms 0.5 seconds 1.0 seconds 1.5 seconds

100 ms 0.7 seconds 1.5 seconds 2.2 seconds

150 ms 0.9 seconds 2.0 seconds 2.9 seconds

200 ms 1.1 seconds 2.6 seconds 3.7 seconds

250 ms 1.3 seconds 3.2 seconds 4.5 seconds

300 ms 1.5 seconds 3.8 seconds 5.3 seconds

350 ms 1.7 seconds 4.3 seconds 6.0 seconds

400 ms 1.9 seconds 4.8 seconds 6.7 seconds

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VXML Call Prompt Setup Times – Call Studio Apps

WAN RTT Latency SIP Call Setup Studio App Download (Initial VoiceXML) Time Until Prompt Plays

50 ms 0.5 seconds 2.0 seconds 2.5 seconds

100 ms 0.7 seconds 3.3 seconds 4.0 seconds

150 ms 0.9 seconds 4.7 seconds 5.6 seconds

200 ms 1.1 seconds 5.9 seconds 7.0 seconds

250 ms 1.3 seconds 7.3 seconds 8.6 seconds

300 ms 1.5 seconds 8.6 seconds 10.1 seconds

350 ms 1.7 seconds 10.0 seconds 11.7 seconds

400 ms 1.9 seconds 11.2 seconds 13.1 seconds

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Understanding Source of Latency Latency Sources

‒ SIP Signaling

Ingress->SIP Proxy->CVP->ICM->VRU Leg = SIP call setup time

‒ Unified CVP Call Server "ping”

Follows VRU Leg, checks to see if CVP is alive

‒ Unified CVP Call root doc fetch

Part of the “ping” includes root doc fetch

‒ Unified CVP VXML Server root doc fetch (Cisco Unified Call Studio only)

Similar to Micro-App however involves Micro-App and Call Studio App

Relatively large transfer and represents the largest amount of delay for the five phases

‒ Cisco Unified Call Studio-based VXML fetch (Cisco Unified Call Studio only)

Follows root doc fetch, represents first interaction with caller

Relatively small fetch resulting in minimal delay

Mitigation Techniques

‒ Reduce WAN RTT latency (Obvious choice)

‒ Inject Audio to “mask” the silence observed by high WAN RTT Latency

‒ Move CVP to a distributed model, placing Call, VXML and Media Servers closer to edge

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Geographic Models

Planning and Design

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Geographic Models

Single Site Advantages ‒ Simplest

‒ Centralized ingress and call control

‒ Ingress and VXML gateways (should) be deployed separately

‒ Dial plan and troubleshooting simplification.

‒ Avoids the need for edge queuing techniques

Disadvantages ‒ Centralized point of failure. Non-flexible

recovery options

‒ Higher equipment costs

Customer Voice Portal (CVP) Geographic Models (1 of 4)

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Geographic Models

Multisite w/ Centralized Call Processing Advantages

Customer Voice Portal (CVP) Geographic Models (2 of 4)

‒ Similar to Single Site advantages

‒ Terminate calls to local agent branches

‒ Ingress and VXML on one gateway at branch locations

‒ Autonomous capacity flex

Disadvantages ‒ Similar to Single Site disadvantages

‒ WAN failure: = device deregistration, loss of call control, queuing, agent state updates, etc.

‒ Possible transcoding and DSP requirements

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Geographic Models

Multisite w/ Distributed Call Processing Advantages

Customer Voice Portal (CVP) Geographic Models (3 of 4)

‒ Similar to Multisite with Central Call Processing

‒ Single site failure redundancy

Disadvantages ‒ Significant solution costs due to infrastructure

replication

‒ Physically diverse networks (private and public) primarily to meet UCCE fault tolerance requirements

‒ Multiple dial plans needed for CUCM

‒ WAN + codec requirements similar to Multisite Central Call Processing

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Geographic Models

Clustering Over WAN Advantages ‒ Most common deployment. Combination of

centralized and distributed call processing.

‒ Fault tolerant and survivability flexible

‒ More efficient use of HW resources

‒ Designed for large enterprises

Disadvantages ‒ Similar to Multisite Distributed

‒ Heaviest dependence on WAN (private and public) availability

Customer Voice Portal (CVP) Geographic Models (4 of 4)

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Branch Design

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

More Edge Queuing Techniques in a moment…

Design for Queuing at the Edge

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Call Admission Control

Why 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

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Branch Design Call 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

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Branch Design Queue 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 Proxy lookup 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

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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

More on how to solve this, stay tuned…

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Edge Queuing Techniques

Planning and Design

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Edge Queuing Set Transfer Label (H.323)

Send to Originator (SIP)

Technology Prefix Stripping (H.323)

Significant Digits (SIP)

Location Based Call Admission Control (LBCAC)

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Send To Originator (SIP) Simple to configure and troubleshoot using OPs Console configuration

Requires that the ingress gateway also provide VXML functionality

SIP Header is checked by CVP SIP service to determine how warm transfer was initiated, avoiding sending VRU label to CUCM.

Best Practice: Always build a centralized VXML router Farm for all transfers when using this technique

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More difficult to configure, requiring pre-pending at ingress gateway

Site Code for VRU leg is pre-pended

Leg Label at Ingress: 228005551212

Site Code must also be embedded for any warm transfers initiated by CUCM, increasing complexity for CUCM dial plan

SIP Proxy dial plan must accommodate switch and VRU labels including site codes

Best Practice: Always build a centralized VXML router Farm for all transfers when using this technique.

Significant Digits (SIP)

22 Site Code for VRU Handling Rest is DNIS for ICM

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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

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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 including warm transfers.

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

Best Practice: Preferred method for SIP edge queuing addressing CAC and deterministic edge queuing for warm transfers. (No need for centralized VXML farm for queuing transfers!)

Location Based CAC Solution (Deterministic Selection of VXML Gateway)

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Ingress and VXML Gateways

High Availability Designs

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Ingress Gateway Configure SIP Server definition via User Agent resolving to DNS

hostname

Configure multiple SIP Proxy dial-peers with same preference for local Data Center proxies.

Configure multiple SIP Proxy dial-peers with higher preference for remote data center. (Yes, higher preference is actually less preferred)

If no SIP proxy server is deployed, setup same dial-peers pointing to Unified CVP Servers.

Consider using DNS SRV values resolved via DNS Server

Consider using DNS SRV values resolved via local static SRV records. (Doesn’t scale and must exist on ALL ingress gateways)

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SIP

High Availability Designs

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SIP Service

SIP is the preferred protocol for CVP

Only SIP (no H323) allowed for new deployments from CVP 8.0

H323 to be totally deprecated in CVP 9.0

SIP provides improved scalability and performance to CVP

SIP and H.323 can co-exist at the same time, however protocol used to initiate the switch leg must remain the same for the entire length of the call! (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

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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.

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Algorithm and Routing Priority

When 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

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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 CVP 8.0 TCP is preferred transport for SIP with CVP. Was UDP prior to CVP 8.0 ‒ CVP running on VMware has issues with UDP

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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

SIP SIP

SIP SIP

ISR

CUCM 73

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SIP Proxy Design

Site A Site B

CUCM Publisher CUCM Subscriber

CTI Custer Aware

ISR ISR

Secondary DialPeer

Primary SIP Trunk Pr

imar

y SI

P Tr

unk

Secondary SIP Trunk

IOS-Based SIP Proxy, Design Option 1 ‒ Single module, very fault tolerant

Cisco Unified SIP Proxy (CUSP) 1. Qualified starting in CVP v7.0(2)

2. CUCM is no longer required

3. SIP module hosted on dedicated Cisco Integrated Service Router (ISR)

Redundancy ‒ Static routes not replicated between sites

‒ SIP option pings can be enabled for CVP network for dynamic CVP state awareness

‒ Option pings at the ingress not yet supported

No Static Route Replication

74

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SIP Proxy Design

Site A Site B

CUCM Publisher CUCM Subscriber

CTI Custer Aware

ISR ISR

Tertiary DialPeer

Tertiary SIP Trunk

IOS-Based SIP Proxy, Design Option 2 ‒ Dual module, very fault tolerant

Cisco Unified SIP Proxy (CUSP) 1. Qualified starting in CVP v7.0(2)

2. CUCM is no longer required

3. SIP module hosted on dedicated Cisco Integrated Service Router (ISR)

Redundancy ‒ Static routes not replicated between sites or between

SIP local SIP Proxies

‒ SIP option pings can be enabled for CVP network for dynamic CVP state awareness

No Static Route Replication

75

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CVP Call Servers

High Availability Designs

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Call and VXML Servers Deploy redundant call servers per data center with sufficient redundant

ports ‒ Note: Redundant CVP port licenses are purchased licenses and live

‒ Note: Gateway/VXML licenses are “paper only” but must be purchased!

Configure SIP Server groups for outbound proxy redundancy

Configure SIP Server groups for dynamic SIP routing and availability

Configure primary and backup VXML Servers on VXML gateway

Consider using content load balancers to load balance VXML and Media Server requests.

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High Availability Design CVP Hardware Redundancy Options

Server Crash

Process Crash/Hang

Performance Bottleneck

Network Outage VVVV

VVVV VVVV VVVV VVVV VVVVVVVV VVVV N + 1 ‒ One additional CVP Call Server

‒ Port redundancy for single server only

‒ Capacity compromised if more than one server is impacted

N + N ‒ Site redundancy without compromising

capacity

‒ Supports maximum hardware, geo-distributed redundancy

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SIP Server Group and Dynamic Routing

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

CVP 8.0

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SIP Server Group

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High Availability Design CVP SRV (Visual)

VVVV

Site A Site B

CUCM Publisher CUCM Subscriber

CTI Custer Aware

SRV Priority 2

VVVV

SRV Priority 3

SRV Priority 2

SRV Priority 3

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Load Balancers

High Availability Designs

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Load Balancers Use to Load Balance HTTP, Media Resource Control

Protocol (MRCP), and RTSP traffic

Cannot load balance call control traffic such as H.323 and SIP

ACE uses a dedicated Fault Tolerant VLAN, CSS uses a VRRP

With ACE, the IP address and MAC address of the FT VLAN do not change during failover

With ACE to achieve active-active redundancy, a minimum of two contexts and two FT groups are required on each ACE.

Best Practice: For larger, geographically separated deployments, deploy pairs of load balancers per data center for localized redundancy.

Caveat: ACE and CSS are the only supported Load Balancer for CVP (F5, etc. are not officially supported)

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ACE Load Balancing Caveat: One arm implementations will break deployments using MRCPv2 since its

based on SIP messages and not RTSP/RTP as with MRCPv1 (RFC 4463).

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Media Servers

High Availability Designs

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Media Servers Configure redundant host entries in VXML gateway

ip host mediaserver 10.1.1.10

ip host mediaserver-backup 10.1.1.11

Keep in mind that “-backup” is only available in Comprehensive Mode ‒ (Anyone know why? Bueller?)

In Standalone mode, use error handling and scripts to pick secondary media server if first fails at VXML gateway

As a fail safe, implement the new “default media server” configuration in Unified CVP.

Best Practice: Always implement a supported load balancer, like ACE or CSS for large, geographically separated media server farms.

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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

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Default Media Server (Cont.)

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) )

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Survivability

High Availability Designs

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Survivability

Allows for a call to reroute during or after system outage

Redirect the caller to alternate destination in the event of either start-of-call or mid-call failures (e.g. ICM Dialogue failure events, failure of IVR leg, media inactivity)

Control recovery behavior based on incoming DNIS and/or time-of-day

Various methods of call recovery:

‒ TDM-only: DTMF*8, Hookflash, TBCT

‒ SIP or TDM:

‒ Hairpin

‒ Transfer to CVP VXML server application

‒ Retry to same DN optionally pre-pended or appended with digit string. Allows finer control over ICM script recovery behavior. For example:

Original DNIS = 8005551212 Recovery DNIS = 999<retry> = 9998005551212

Works in CVP Standalone or Comprehensive Models

Requires ASR1000, CUBE or TDM gateway

Cannot recover call context from the original call

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CVP Survivability

Caller (TDM or IP)

CVP Call Server

Voice Gateway

VXML Gateway

X

SIP

GED-125

RTP

Upon receiving failure notification from ICM or VXML gateway, Call Server sends BYE with abnormal cause code which causes gateway to invoke survivability.

Alternately, if the gateway detects a loss of media to an IP phone, it can invoke survivability if media inactivity timers are set.

Survivability

X

X

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What happens if you fail to plan?

Case Studies

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“Better Late than Never” The Situation:

Customer designed and deployed a centralized Unified CCE solution using the “Clustering over the WAN” deployment model with SIP and edge queuing. Branch locations were located in UK, Germany, Hong Kong, China, etc.

An international carrier was picked to provide for local and international dialing services as well as TFN ingress services at the branch. Unified Call Studio was deployed to build self service applications as well as Collect and Prompt services on servers in the US.

Progression:

US Based branches were deployed without any issues

UK and Germany based branches were deployed with minor issues and complaints about prompt delays. This was determined to be ~2-3 seconds longer than previous system.

APAC deployment suffered from massive delays from the time the call was placed to when the caller heard a prompt.

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“Better Late than Never” The Issue:

US Based branches had low latency WAN connections to their branches <50 ms

UK and Germany based branches had higher than normal latency than the US, placing them around 100 to 150 ms, still acceptable from the business perspective.

APAC branches suffered from massive latency ranging anywhere from 300 ms to 400 ms RTT (10-13 second addition to overall prompt delivery).

It was discovered that APAC branches were also suffering from a carrier connect delay of 5-6 seconds.

Resolution and Lesson:

Customer eventually had to move CVP to a distributed model, placing call, VXML servers, SIP Proxy, etc. into a regional hub where latency to APAC branch offices was greatly reduced.

Know your latency numbers and design accordingly! Distributing CVP after the fact is expensive and a painful process.

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“Trust but Verify…” The Situation:

Customer designed and deployed a centralized Unified CCE solution using the “Clustering over the WAN” deployment model with SIP and edge queuing. Branch locations were located throughout the US with CVP located in two US based data centers.

Progression:

US Based branches deployed initially without any serious issues

Initial deployment of live agents was minor in size.

Within 2 months, almost 2K agents were placed on the solution

Slowly the quality of call treatment transactions and agent calls degraded over time to a point of failure.

Customer’s voice and network teams defended their network, their QoS policies and implementations with vigor…

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“Trust but Verify…” The Issue:

It was discovered that why the customer had a QoS policy and template, its implementation from the LAN and WAN perspective were not handled by the same group nor were they consistent and correct.

Traffic was being marked close to the source, however, not trusted as it crossed into the WAN realm, essentially unmarking it.

Furthermore, HTTP traffic, critical to CVP VXML instructions, was being marked as scavenger traffic and drop due to contention of bandwidth.

Customer also didn’t have accurate bandwidth utilization numbers for WAN connections to the branches prior to implementation.

Resolution and Lesson:

Trust that your customer has QoS implemented, Verify end to end that it is implemented and honored correctly.

Know the bandwidth consumption of a branch, especially if you are adding agents and voice services.

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New Unified CVP Book

Title: Cisco Unified Customer Voice Portal: Building Unified Contact Centers

ISBN-10: 1-58714-290-2 and ISBN-13: 978-1-58714-290-1

Authored by: Rue Green

Grab a copy at the Cisco Book Store

Published December 2011 by Cisco Press

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Complete Your Online Session Evaluation Give us your feedback and you

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Complete your session evaluation online now (open a browser through our wireless network to access our portal) or visit one of the Internet stations throughout the Convention Center.

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IP communications, messaging, mobile applications, telepresence

Interact with Cisco Product Managers, Technical Marketing Engineers and Services Consultants

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Join the Collaboration User Group

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Access to early product releases and info

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Visit the Cisco Collaboration Booth (#1289) at Cisco Live: Learn more about the community. Sign up for the user group. Pick up your polo shirt and badge ribbon (user group members only).

communities.cisco.com

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Final Thoughts

Get hands-on experience with the Walk-in Labs located in World of Solutions, booth 1042

Come see demos of many key solutions and products in the main Cisco booth 2924

Visit www.ciscoLive365.com after the event for updated PDFs, on-demand session videos, networking, and more!

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