SIP-Based Architectures for Cisco Contact Center Solutions and Collaboration

BRKCCT-2030
Dino Romero

Agenda
Introduction & Scope Architecture Overview
Carrier Connectivity & Call Control SIP Call Flows

Customer Case Study Migration Strategy Architectural End State

Additional Documentation & Reference Material Q&A
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Scope of this Session

CVP

A
A

CVP

SP SIP
Campus

Contact Centers CVP A

Branch Offices

Learn how to future proof your Contact Center and Collaboration Architecture with SIP Understand migration strategies from legacy to prevailing SIP designs in Contact Center and Collaboration
Learn First hand customer experiences and best practices during a SIP migration
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Outside of the Scope of this Session

This session is not a deep dive design review This session does not review every migration possibility and/or IPto-IP call flow scenario

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General Business Drivers for SIP Adoption

The overall industry trend is moving towards using SIP for call signaling
Increased feature velocity and 1st contact resolution (Courtesy call back, better end-to-end customer experience)

Future R&D is SIP focused

Ease of interoperability (applications and multi-vendor)

SIP end-to-end strategies (session management, policy management, call control, routing, application layer integration, etc.)

Migration to SIP carrier trunking and consolidating carrier connectivity

Significantly easier to scale carrier trunking capacity Rapid response to call volume changes Cost Savings
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Architecture Overview

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Key Factors Impacting Architecture

Hosted or premise based solution

Partner/Outsourcer integration Centralized vs. decentralized Regardless of design (centralized, decentralized, hybrid) the components are the same. Only difference is where components are physically deployed.

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High Level Architecture

IP-Carrier
CUBE CUBE CUBE

IP-Carrier

Contact Center Data Center A

CUBE

PG PG

PG PG
NIC-B

PG PG

NIC-A

(VoIP) WAN
Contact Center

Private Link
Data Center B

PG PG

PG PG
NIC-A

PG PG

Partner/Outsourcer Contact Center

SIP Trunking
CUBE

NIC-B
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Carrier Connectivity & Call Control

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Carrier Connectivity The Basics

Centralized vs. Decentralized

Delivery Mechanism (i.e. Ethernet, T1, DS3)

Session Border Control (SBC) Security

Carrier

CUBE

Gateways
Carrier

CUBE

Inbound vs. Outbound

SIP and/or TDM?

Carrier Features
TBNT SIP Refer
Mid-Call Codec Renegotiation

BRKUCC-2006: SIP trunk design and IP PSTN Trunking Additional Sessions

BRKUCC-2006: SIP trunk design and deployment BRKUCC-2735: SIP Trunk Design and Deployment Playbook for the Enterprise BRKUCC-2403: Understanding the Intercompany Media Engine solution BRKUCC-2931: Case Study for Large Scale Centralized SIP Trunk implementation LTRUCC-2150: Cisco Unified Communications SIP Trunking, Session Management, and Service Advertisement Framework

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Enterprise Feature Considerations

Transcoding/Transrating

Message Normalization/SIP Profiling Protocol & DTMF Interoperability High Availability Requirements (i.e. Site Survivability Requirements)
Recording (Media Forking) Call Admission Control Dial PlanData Center A Intelligence
PG PG PG PG
NIC-B
CUBE

Security NAT Firewalling Secure RTP Antivirus/Malware Protection (CSA)

BRKCCT-1041 Changing the way we record voice and video on the network with Cisco MediaSense BRKCDN-1102 Programming with the MediaSense API

Contact Center

(VoIP) WAN
PG PG
Contact Center

NIC-A

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Call Control

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Call Control Elements Core

SIP Proxy Options
Cisco Unified Sip Proxy (CUSP)
Integrated Solution in ISR Gateway Call Stateless Cisco Unified Presence Server (CUPS) Server based
BRKUCC-1305 Understanding the Session Border Controller Portfolio from Cisco BRKUCC-2785 Understanding Cisco Unified Presence 8.x LTRUCC-2150: Cisco Unified Communications SIP Trunking, Session Management, and Service Advertisement Framework

CUSP

Newest SIP Proxy commonly used with 8.x

Typically used with 7.5 or earlier deployments Call Stateless

CUPS

Cisco Unified Communications Manager Session Management Edition (SME)

Functions as a Back to Back User Agents (B2BUA) Based on CUCM OS, Part of CUCM cluster
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Call Control Elements Core

Comparison

Cisco Unified SIP Proxy (CUSP) Improved scalability No communication/call memory structure to be maintained Resiliency - (Record Route Off - subsequent SIP messages do not flow through CUSP) Header manipulation Session Manager Edition (SME) Feature interworking (i.e. DTMF interworking Ability to invoke mid-call features Reduced scalability Redundancy for New Call Setup Header manipulation Transcoding

CUSP

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Call Control Elements Advanced

Session Border Controller (SBC)
CUBE

Cisco Unified Border Element (CUBE)

Voice XML, Self-Service, and Queuing

Cisco Voice Portal (CVP) Comprehensive Contact control Functions as a Back to Back User Agents (B2BUA) Contact Center Queuing, Routing, Reporting, etc.

Unified Contact Center Enterprise (UCCE)

Logical Queue & resource selection

Unified Communications Manager

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SIP Call flows

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SIP Call Flows Assumptions

Component Assumptions Product Release

CUBE CVP ICM

CUCM

15.1(2T)=8.5 8.0(2) Compatibility Matrix

8.0(2)

The following call flows assume the above versioning at (a minimum)

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SIP Call Flows

Ingress (from Carrier)

Enterprise Interior Best Practices/Recommendations

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Basic SIP Call with Cisco UCCE and CVP

G.729/G.711

VXML Gateway

7 2 3 4 6
8

SIP

1
G.729/G.711

CUBE Does not require DSPs

8
8

VXML Service Control SIP RTP JTAPI Call Control

Contact Center

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Basic SIP Call with Mixed Codec

G.711

VXML Gateway

SIP

G.711/G.729 CUBE

G729 provided SP supports mid-call codec renegotiation (likely & preferred) G711 if SP does not support mid-call codec renegotiation requires DSPs on CUBE

G.729

VXML Service Control SIP RTP JTAPI Call Control

Contact Center

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Warm Transfer to CVP (from Agent)

VXML Gateway

SIP

G.729/G.711

CUBE Does not require DSPs

G.729/G.711

1
VXML Service Control SIP RTP JTAPI Call Control

Contact Center

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Warm Transfer to CVP Continued

VXML Gateway

SIP

G.729/G.711

CUBE Does not require DSPs

G.729/G.711 Call on Hold

3
Agent initiated transfer

G.729/G.711

Contact Center

VXML Service Control SIP RTP JTAPI Call Control

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Transfer to CVP Complete (Call in Queue)

G.729/G.711

VXML Gateway

6
CUBE

SIP

G.729/G.711

7 8

Does not require DSPs

Contact Center

VXML Service Control SIP RTP JTAPI Call Control

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Transfer to CVP Mixed Codec

VXML Gateway

SIP
G729 provided SP supports mid-call codec renegotiation (likely & preferred) G711 if SP does not support mid-call codec renegotiation requires DSPs on CUBE

G.711

CUBE Does not require DSPs

G.729

1
VXML Service Control SIP RTP JTAPI Call Control

Contact Center

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Agent Warm Transfer to Queue

VXML Gateway

SIP

G729 (or G711 CUBE if SP doesnt Does not require DSPs support midcall renegotiation)

G.729 Call on Hold

**Note mixed codecs required for agent Agent-VXML re-queue using G711 because prompts are G711 for speech portion

3
Contact Center

G.711

VXML Service Control SIP RTP JTAPI Call Control

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Agent Warm Transfer to Queue

VXML Gateway

SIP

G729 (or G711 CUBE if SP doesnt Does not require DSPs support midcall renegotiation)

4
G.729
DSPs required for transcoding

**CUCM disengages DSPs while call in queue

Contact Center

G.711

VXML Service Control SIP RTP JTAPI Call Control

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Agent Warm Transfer to Queue (Complete)

G.729/G.711

VXML Gateway

CUSP

6 7 8

SIP

G.729/G.711

CUBE

Does not require DSPs

Contact Center

VXML Service Control SIP RTP JTAPI Call Control

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Best Practices
Use flat codecs if possible to minimize complex DSP sizing G711 Preferred If mixed codecs are used, detailed call flows will be required to understand DSP needs. Most common driver for mixed codecs is when speech recognition AND remote locations (ie., using G729) are present at the same time Avoid G729 on the SP when speech is used (i.e. dont try to transcode the call to G711, since call quality is already reduced) Understand SP capabilities to do mixed-codec (re)negotiation (may drive the need for more DSPs in some call flows)
Use single encoding for CVP prompts (G711), even in mixed codec environments

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Best Practices - Continued

Understand SP/carrier capabilities for TBNT, especially how REFERs are handled from the enterprise back into the carrier network Do careful capacity engineering of all the components (i.e. conferencing, transfers & DSPs)

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Customer Case Study

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Customer Case Study

Global enterprise software company North America serves 3.1 million SMEs, healthcare and specialized industries
Operate 26 contact centers distributed across North America Avaya, Aspect and Nortel served as voice and ACD 1,500+ agent resources (inbound and outbound) Contact Centers include sales and support for software products

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Customer Case Study continued

Phase 1 moved to SIP trunks
Deployed Cisco CUBE as SBC in 2 data centers Utilized MPLS to back haul voice to contact center Deployed Cisco ISR for voice gateway at the contact center
ACD PRIs were moved to Cisco ISR at each contact center Reduced carrier PRIs at site (largest 53 PRIs)

Lessons Learned
Major cost reduction main driver

Do not assume feature and number availability (consolidated DID range) Long lead time on new DID ranges Maintenance Windows were not long enough (after pre-port numbers were stopped in working hours)
Fax would recommend as a standalone project (carrier did not support T.38)

Outbound calls required authentication with owed number

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Customer Case Study continued

Lessons Learned
Call paths purchased were not pooledbe sure to verify with carrier Configured DSP with medium complexity using g729r8

Phase 2
Replace legacy ACDs with Cisco UCCE and CVP - ingress

Lessons Learned
g729 licensed codec requires conversion of recorded prompts SIP rel1XX is not currently supported with Cisco CVP (removed from dial peers)

Carrier could not provide DID in Canada

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Migration Strategy

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Migration Strategy
Migration from Legacy Platforms

H.323 (Gatekeeper) to SIP Migration Hosted IVR and Pre-Routing SIP Migration Best Practices & Recommendations

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Migration from Legacy Platforms

Migration to VoIP is well underway
Toll-Bypass including elimination of Take Back & Transfer (TBNT) fees & Carrier NIC fees
End-to-end call/contact control (i.e. elimination of hairpinning calls) Velocity of features (i.e. Service Provider & Enterprise mid-call codec renegotiation, Courtesy Call Back, etc.)
Email

Phone

SIP to Simplify
First Call/Contact Resolution (i.e. expert availability)

All-in-one Framework SIP has emerged as the universal blueprint for IP Telephony & Unified Communications

SIP

Chat

Begin with End in Mind

Understanding desired End-State Architecture
Developing concept of phased migration to End-State Leverage existing Assets (ICM is foundation)

Web Portal

Video Content

Easily layer on new capabilities (i.e. Voice, web, Email, SMS, Voice recording)
Kiosk

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H323 (Gatekeeper) to SIP Migration

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H.323 Gatekeeper Routing

Carrier Carrier

Gatekeeper
ICM ICM

Gatekeeper

Private Link

Data Center A

Data Center B

Phones

VoIP/DATA WAN
Phones

Agents

PG

Agents

PG

Contact Centers
Phones

Contact Centers
Phones

Agents

PG

Agents

PG

Queuing Locally
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Queuing Locally
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H.323 Gatekeeper Routing Continued

Carrier Carrier

Centralized Queueing
CUBE CUBE
ICM ICM

H323 Component Removal including RAS Dial Peers on Gateways

Gatekeeper

PG

ICM ICM

PG

Gatekeeper

Private Link

Data Center A

Data Center B

CUSP adds dial plan centralization

Phones

VoIP/DATA WAN
Phones

Agents

PG

Agents

PG

Contact Centers
Phones

Agents

PG

BRKCCT-3007: Upgrade Planning for Cisco Unified Intelligent Contact Management and Cisco Unified Customer Voice Portal BRKUCC-1021: Cisco Unified Customer Voice Portal and Video Contact Center Update and Roadmap BRKUCC-2020: Planning and Designing a Cisco Unified Customer Voice Portal Deployment
2011 Cisco and/or its affiliates. All rights reserved.

Contact Centers
Phones

Agents

PG

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H.323 to SIP Migration

TDM

G7XX

Gatekeeper

ICM

TDM

Media (RTP) SIP Call Control

RAS/GKTMP

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H.323 to SIP Migration Continued

TDM G7XX Does not require DSPs

CUBE

G7XX

G7XX

Gatekeeper

TDM

Hybrid Environment ICM & UCCE Scripts

VXML Gateway

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H.323 to SIP Migration Continued

TDM G7XX Does not require DSPs

CUBE

G7XX

G7XX

Hybrid Environment ICM & UCCE Scripts

VXML Gateway

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Hosted IVR & Pre-Routing SIP Migration

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Hosted IVR and Pre-Routing

Carrier Carrier

ICM

ICM

Private Link
NIC-B Data Center B

NIC-A Data Center A

Phones

VoIP/DATA WAN
Phones

Agents

PG

Agents

PG

Contact Centers
Phones

Contact Centers
Phones

Agents

PG

Agents

PG

Queuing Locally
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Hosted IVR and Pre-Routing Continued

Carrier Carrier

PG
ICM ICM

PG Private Link
NIC-B Data Center B

NIC-A Data Center A

Phones

VoIP/DATA WAN
Phones

Agents

PG

Agents

PG

Contact Centers
Phones

Contact Centers
Phones

Agents

PG

Agents

PG

Queuing Locally
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Hosted IVR and Pre-Routing Continued

Carrier

Centralized Queuing

Carrier

PG

CUBE
ICM

CUBE

PG
ICM

Private Link

Data Center A

Data Center B

Phones

VoIP/DATA WAN
Phones

Agents

PG

Agents

PG

Contact Centers
UCCE
Phones

Contact Centers
Phones

Agents

PG

Agents

PG

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Hosted IVR & NIC SIP Migration

TDM

TDM Hosted IVR

ICM
TDM TBNT

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Hosted IVR & NIC SIP Migration Continued

G7XX Does not require DSPs TDM

TDM Hosted IVR

CUBE

TDM TBNT

VXML Gateway

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Hosted IVR & NIC SIP Migration Continued

G7XX Does not require DSPs

CUBE

VXML Gateway

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3rd Party SBC to CUBE

IP Carrier
Non-contact Center SIP Traffic
SBC

Enterprise SBC
SBC
Contact Center SBC CUBE CUBE
ICM

IP Carrier

PG

PG
CUBE CUBE
ICM

CUBE

Data Center A

Private Link

Data Center B

Phones

VoIP/DATA WAN
Phones

Agents

PG

Agents

PG

Contact Centers
Phones

Contact Centers
Phones

Agents

PG

Agents

PG

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Best Practices & Recommendations

Deploy the Cisco Unified Border Element (CUBE) in your Collaboration IP-to-IP architectures CUBE helps interconnect these networks that are in various stages of transition

CUBE TCL/VXML resiliency

Ensures QoS PSTN (TDM) Fallback for Short term Call Admission Control (CAC) to limit call arrival rates & max active calls

Deploy The Cisco Unified SIP Proxy (CUSP) in your Collaboration IP-to-IP architectures for the replacement of H323 dial plans and/or for introducing Scalable SIP centralized dial plans.

Use a B2BUA (SME) to do rich feature interoperability, mid-call feature invocation and/or sophisticated SIP header manipulation for more complex or irregular call flows
Determine the migration steps of your contact center and whether it is best for your business if they happen all at once, or in sequence
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Architectural End state

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Phase 0

Architectural Migration
Carrier TDM

TB&T through TDM cloud

Carrier TDM

Gatekeeper
ICM ICM

Gatekeeper

PG
Data Center A

Private Link

PG
Data Center B Outsource Providers

Outsource Providers

Enabled via Corporate UCCE

Enabled via Corporate ICM

IP WAN
Agents

CAD/CTIOS

Finesse

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Agents

Agents

Agents

Agents

Agents

Agents

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Phase 1

Architectural Migration Continued

Carrier TDM Carrier TDM

Enterprise Queue & IVR

Gatekeeper

Email & CVP Web Interaction

ICM ICM

Email & CVP Web Interaction

Gatekeeper

PG
CUPS

Private Link

PG
CUPS

Data Center A

Data Center B Outsource Providers

Outsource Providers

Enabled via Corporate UCCE

Enabled via Corporate ICM

IP WAN
Agents

CAD/CTIOS

Finesse

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Agents

Agents

Agents

Agents

Agents

Agents

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Phase 2

Architectural Migration Continued

Carrier IP Carrier IP

Enterprise Queue & IVR

Recording (Media Sense) CUSP

Email & CVP Web Interaction

Outbound SIP Dialer

CUBE
ICM

CUBE
ICM

Outbound SIP Dialer

Email & CVP Web Interaction

Recording (Media Sense)

PG
Social Media (Standalone)

Private Link
Social Media (Standalone)

PG

CUSP

Data Center A

Data Center B Outsource Providers

Outsource Providers

Enabled via Corporate UCCE

Enabled via Corporate ICM

IP WAN
Agents

CAD/CTIOS

Finesse

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Cisco or non-Cisco Phones

Agents

Agents

Agents

Agents

Agents

Agents

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Phase 3A

Architectural Migration Continued

Contacts
PSTN/Internet

Contacts

SIP IP Access Access

Services Advertisement Framework (SAF)

SIP HTTP Social Media (Blogging)

Presence XMPP (Jabber)

SME

Recording (Media Sense)

Video

XML Routing

SIP

HTTP

Services Advertisement Framework (SAF)

User

LTRUCC-2150: Cisco Unified Communications SIP Trunking, Session Management, and Service Advertisement Framework

3rd Party

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Phase 3B

Architectural Migration Continued

Contacts
PSTN/Internet

Contacts

SIP
IP Access

Access

Services Advertisement Framework (SAF)

Social Miner (Blogging) HTTP

Presence XMPP (Jabber)

Shared Applications
HTTP

SME
SIP

Video
SIP

SME

Recording (Media Sense)

XML Routing

Services Advertisement Framework (SAF)

User

3rd Party

BRKCCT-2030

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Additional Documentation & Reference Material

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Related Sessions
Customer Collaboration Technologies Sessions
BRKCCT-1021: Cisco Unified Customer Voice Portal and Video Contact Center Update and Roadmap BRKCCT-1041 Changing the way we record voice and video on the network with Cisco MediaSense BRKCDN-1102 Programming with the MediaSense API BRKCCT-2007: Planning and Designing a Cisco Unified Contact Center Enterprise Deployment BRKCCT-2020: Planning and Designing a Cisco Unified Customer Voice Portal Deployment BRKCCT-3007: Upgrade Planning for Cisco Unified Intelligent Contact Management and CVP TECCCT-2002: Cisco Planning and Designing a Cisco Unified Contact Center Enterprise Deployment TECCCT-2022: Customer Voice Portal Technical Seminar: Planning, Design and Troubleshooting

IP PSTN Trunking
BRKUCC-2006: SIP trunk design and deployment BRKUCC-2735: SIP Trunk Design and Deployment Playbook for the Enterprise BRKUCC-2403: Understanding the Intercompany Media Engine solution BRKUCC-2931: Case Study for Large Scale Centralized SIP Trunk implementation LTRUCC-2150: Cisco Unified Communications SIP Trunking, Session Management, and Service Advertisement Framework

Pltforms

TECARC-2002: ISR G2 Fundamentals BRKARC-3001: Cisco Integrated Services Router G2 Architectural Overview and Use Cases BRKARC-2001: Cisco ASR1000 Series Routers: System and Solution Architectures
BRKUCC-2002: Securing Enterprise Voice Network and Applications BRKUCC-3501: CUCM Security BRKUCC-2003: A new approach to call routing and dial plans based on the Service Advertisement Framework (SAF) BRKVVT-2010: Designing UC Gateways and DSP Engineering in Enterprise Networks BRKUCC-2012: Understanding SIP Endpoints in Cisco Unified Communications Manager
2011 Cisco and/or its affiliates. All rights reserved.

Security
Miscellaneous

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Cisco.com SIP Trunk and CUBE Resources

Cisco CUBE on Cisco.com
http://www.cisco.com/go/cube

Cisco Communications Transformations Whitepapers

http://www.cisco.com/go/cube (section on Whitepapers)

Cisco UBE SP SIP Trunk Interoperability Reports

www.cisco.com/go/interoperability > Cisco Unified Border Element (CUBE)/SIP Trunking Solutions

Cisco UBE PBX Interoperability Reports (Avaya/Nortel)

www.cisco.com/go/interoperability > Cisco Unified Border Element (CUBE)/SIP Trunking Solutions

CUCM SIP Trunk Documentation: www.cisco.com/go/srnd

8.x: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1044916 7.x: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1044916
6.x: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/trunks.html#wp1044916

CVP 7.0 SIP Trunk Integration: www.cisco.com/go/srnd

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/customer_voic e_portal/srnd/7x/cvp_gtwy.html#wp1048342
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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
SIP Trunking [1-58705-9444] Cisco Voice Gateways and Gatekeepers [1-58705-258X] 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, scheduled to be published late 2011

Available Onsite at the Cisco Company Store

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Q&A

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Complete Your Online Session Evaluation

Receive 25 Cisco Preferred Access points for each session evaluation you complete. Give us your feedback and you could win fabulous prizes. Points are calculated on a daily basis. Winners will be notified by email after July 22nd.

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. Dont forget to activate your Cisco Live and Networkers Virtual account for access to all session materials, communities, and ondemand and live activities throughout the year. Activate your account at any internet station or visit www.ciscolivevirtual.com.

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Visit the Cisco Store for Related Titles http://theciscostores.com

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BRKCCT-2030

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Thank you.

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Additional Reference Slides

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DTMF Interworking Support

H.323 H.323
H.323
H.245Alphanumeric H.245-Signal RFC2833

H.323 SIP
H.323
H.245Alphanumeric

SIP SIP
SIP SIP

H.323
H.245Alphanumeric H.245-Signal RFC2833

SIP
NOTIFY

NOTIFY RFC2833
RFC2833 KPML Voice InBand*

NOTIFY NOTIFY
RFC2833 KPML

H.245-Signal RFC2833
H.245Alphanumeric H.245-Signal RFC2833

NOTIFY NOTIFY RFC2833

RFC2833 RFC2833 KPML KPML RFC2833
Cisco Public

H.245Alphanumeric
H.245-Signal Voice In-Band*

RFC2833
RFC2833 RFC2833

RFC2833

All DTMF Interworking Is Bidirectional

*Requires Transcoder DSP
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H.245alphanumeric
H.245-Signal Voice In-Band*

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Call Admissions Control

CUBE provides various different CAC mechanisms
Total calls, CPU, Memory, GK IP call capacity, max-connections, RSVP

Total Calls, CPU, Memory

CUBE

High Water Mark Low Water Mark

call threshold global [total/mem/cpu] calls low xx high yy

Call Spike Detection

CUBE

call spike call-number [steps number-of-steps size milliseconds] call spike 10 steps 5 size 200

If a call spike is detected, reject calls

Max Calls per Destination

Call #1
Call #2 Call #3
CUBE

dial-peer voice 1 voip max-conn 2

Call #3 Rejected by CUBE

Cisco Public

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SIP Trunk Best Practices

The SIP trunk market is maturing Use CUBE as the onsite enterprise
Plan and execute thorough testing before production Most interop issues can be resolved with targeted configuration changes and protocol normalization Border Element to SIP DO-EO conversion Normalize traffic SP UNI Interconnect/share a SIP trunk to different enterprise IP-PBXs Security for CUCM/enterprise apps QoS and troubleshooting demarc Use a G.711 SIP trunk Avoid transcoding if possible Ensure these are addressed: Redundancy especially for large, centralized SIP trunk designs Fax Emergency Calls DID porting
Cisco Public

Evaluate different providers

Offerings vary considerably

CUCM recommendations
CUCM 5.x and older: H.323 CUCM 6.x and newer: SIP Avoid MTPs if possible

Use the SRNDs and Configuration App Notes

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Basic B2BUA Call Flow Why CVP and UCM (and others) are a B2BUA
SIP UA-1
INVITE B2BUA

SIP UA-2 INVITE 200 ACK BYE 200 BYE 200 INVITE
Major difference between B2BUA and Proxy. Proxy cant do this

200 ACK BYE 200

And here as well

SIP UA-3

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Basic SIP Call Flow

Record Route is disabled for faster failover

CVP
CVP VXML GW ICM Egress VGW/ UC Manager IP PHONE

Session Controller

Invite

SIP Proxy Invite Invite

GED-125 new call

Send to VRU to Queue call using VXML GW label

Invite
Invite

alerting

200 OK
HTTP New Call

200 OK 200 OK

GED-125 200 OK
play script
Do we need to describe this better? Connect/ Answer

VRU leg request instruction. Micro app plays prompts & collects digits

2-way RTP

HTTP vxml IOS VB executes vxml scripts

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Basic SIP Call Flow (part 2)

Session Controller
Flow-Through

SIP Proxy RTP

Record Route is disabled for faster failover

CVP
CVP VXML GW ICM

Egress VGW/ UC Manager

IP PHONE

Caller O HTTP result

Only a B2BUA can do this

GED-125 script results

agent label
BYE 200 OK Invite

Connect / Transfer to agent or queuing then connect to agent

Invite
200 OK 200 OK ACK Re-Invite

Ring MGW

In SIP domain, ringing is at the originating endpoint. Since the call has been established for vxml session, CVP needs to connect to GW for ring tone to caller while waiting for agent transfer completed

200 OK
ACK
RTP Tcl plays ringing tone
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Basic SIP Call Flow (part 3)

Record Route is disabled for faster failover

CVP
CVP VXML GW
ICM Either Egress VGW or UC Manager IP PHONE

Session Controller
Flow-Through

2-way RTP

SIP Proxy

Invite Invite

Ring MGW

SIP/SCCP

200 OK
200 OK BYE 200 OK Re-Invite 200 OK ACK ACK
Flow-Through 2-way RTP

Answer

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Release Port IP Transfer (SIP Refer)

Session Controller
Flow-Through

SIP Proxy
RTP

Record Route is disabled for faster failover

CVP
CVP VXML GW ICM

Egress VGW/ UC Manager

IP PHONE

HTTP result

Caller O
Connect / Transfer to agent using Refer Label

GED-125 script results Refer Label Z

BYE 200 OK
Refer (agent label Z)

202 Accepted

CVP will be out of the loop

Invite

Invite
180 ringing 180 ringing Notify ringing 200 OK 200 OK Notify 200 BYE
CVP is out of the call, port license is released answer

SIP/SCCP

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Release Port 302 Transfer Immediate

Record Route is off for faster failover

CVP
CVP VXML GW ICM

Session Controller
Invite

Invite

SIP Proxy Invite (SDP-a)

Egress VGW/ UC Manager

IP PHONE

GED-125 script results Refer Label Z 302 (label) 302 (label)

Dialog is not established, use 302 Redirect; not Refer
Connect / Transfer to agent using Refer Label

ACK
Invite

Invite Z 180 ringing 180 ringing

Notify ringing 200 OK 200 OK Notify 200 OK BYE
CVP is out of the call, port license is released

SIP/SCCP

answer

ACK

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Release Trunk TDM Transfer (SIP Info)

Record Route is off for faster failover

CVP
CVP VXML GW

Session Controller
Flow Through

SIP Proxy RTP

ICM

Egress VGW/ UC Manager

IP PHONE

HTTP result

Caller O Connect / Transfer to agent using Refer Label replacement to DTMF *8 Label and ICM translation route to preserve call context. ECC UUI is optionally set for outbound delivery

GED-125 script results **Refer Label**

RTP NTE BYE

Based on RFC 2833 NTE (DTMF) digits are inband

200 OK

CVP is out of the call

BYE (GTDs UUS) BYE) 200 OK 200 OK

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