BRKCCT-2030

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BRKCCT-2030

SIP-Based Architectures forCisco Contact Center Solutions

and Collaboration

Dino Romero

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2011 Cisco and/or its affiliates. All rights reserved. Cisco PublicBRKCCT-2030 2

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

Learn how to future proofyour Contact Center and CollaborationArchitecture with SIP

Understand migration strategies from legacy to prevailing SIPdesigns in Contact Center and Collaboration

Learn First hand customer experiences and best practices during a

SIP migration

A

A

CVP

SP SIP

Campus

Contact Centers

Branch Offices

A

CVP

CVP

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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 IP-to-IP call flow scenario

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The overall industry trend is moving towards using SIP forcall 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 carrierconnectivity

Significantly easier to scale carrier trunking capacity

Rapid response to call volume changes

Cost Savings

General Business Drivers for SIP Adoption

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Hosted or premise based solution

Partner/Outsourcer integration

Centralized vs. decentralized

Regardless of design (centralized, decentralized, hybrid) thecomponents are the same. Only difference is where components arephysically deployed.

Key Factors Impacting Architecture

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High Level ArchitectureIP-Carrier

(VoIP) WAN

Private Link

IP-Carrier

NIC-B

NIC-A

Data Center A

CUBECUBECUBE

PGPG

PGPG

Contact Center

PGPG

NIC-A

NIC-B

Data Center B

PGPG

PGPG

CUBE

Contact Center

PGPG

Partner/Outsourcer Contact Center

CUBE

SIP Trunking

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Carrier Connectivity & Call Control

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Centralized vs. Decentralized

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

Session Border Control (SBC)

Security

Gateways

Inbound vs. Outbound

SIP and/or TDM?

Carrier Features

TBNT

SIP Refer

Mid-Call Codec Renegotiation

Carrier Connectivity The Basics

Carrier

CUBE

Carrier

CUBE

BRKUCC-2006: SIP trunk design and IP PSTN TrunkingAdditional Sessions

BRKUCC-2006: SIP trunk design and deployment

BRKUCC-2735: SIP Trunk Design and Deployment Playbookfor the Enterprise

BRKUCC-2403: Understanding the Intercompany MediaEngine solution

BRKUCC-2931: Case Study for Large Scale Centralized SIPTrunk implementation

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

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Transcoding/Transrating

Message Normalization/SIP Profiling

Protocol & DTMF Interoperability

High Availability Requirements (i.e.

Site Survivability Requirements)

Recording (Media Forking)

Call Admission Control

Dial Plan Intelligence

Enterprise Feature Considerations

Security

NAT

Firewalling

Secure RTP

Antivirus/Malware

Protection (CSA)

NIC-B

NIC-A

Data Center A

CUBE

PGPG

PGPG

Contact Center

PGPG

(VoIP) WAN

Contact Center

BRKCCT-1041 Changing the way werecord voice and video on the networkwith Cisco MediaSenseBRKCDN-1102 Programming with theMediaSense API

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Call Control

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SIP Proxy Options Cisco Unified Sip Proxy (CUSP)

Integrated Solution in ISR Gateway

Newest SIP Proxy commonly used with 8.x

Call Stateless

Cisco Unified Presence Server (CUPS)

Server based

Typically used with 7.5 or earlier deployments

Call Stateless

Cisco Unified Communications Manager SessionManagement Edition (SME)

Functions as a Back to Back User Agents (B2BUA)

Based on CUCM OS, Part of CUCM cluster

BRKUCC-1305 Understanding theSession Border Controller Portfolio

from CiscoBRKUCC-2785 Understanding CiscoUnified Presence 8.x

LTRUCC-2150: Cisco UnifiedCommunications SIP Trunking,Session Management, and ServiceAdvertisement Framework

Call Control Elements Core

CUSP

CUPS

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Comparison

Cisco Unified SIP Proxy (CUSP)

Improved scalability No communication/call

memory structure to be maintained

Resiliency - (Record Route Off - subsequent

SIPmessages 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

Call Control Elements Core

CUSP

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Session Border Controller (SBC)

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

Call Control Elements Advanced

CUBE

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SIP Call flows

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Component Assumptions

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

SIP Call Flows Assumptions

Product Release

CUBE 15.1(2T)=8.5

CVP 8.0(2)ICM Compatibility Matrix

CUCM 8.0(2)

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Ingress (from Carrier)

Enterprise Interior

Best Practices/Recommendations

SIP Call Flows

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SIP

Basic SIP Call with Cisco UCCE and CVP

5

2

7

4

Contact Center

VXMLGateway

1 6

VXMLService Control

SIP

RTPJTAPI

Call Control

Does not require DSPs

G.729/G.711

G.729/G.711

8

CUBE

8

8

3

7

7

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SIP

Contact Center

VXML

Gateway

VXMLService Control

SIP

RTPJTAPI

Call Control

G.711

G.711/G.729CUBE

G729 provided SP supportsmid-call codec renegotiation

(likely & preferred)

G711 if SP does not supportmid-call codec renegotiationrequires DSPs on CUBE

G.729

Basic SIP Call with Mixed Codec

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SIP

Contact Center

VXML

Gateway

CUBE

1

VXMLService Control

SIP

RTPJTAPI

Call Control


G.729/G.711

G.729/G.711

Warm Transfer to CVP (from Agent)

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Warm Transfer to CVP Continued

SIP

2

Contact Center

VXML

Gateway

CUBE

VXML

Service Control

SIP

RTPJTAPI

Call Control


G.729/G.711

G.729/G.711Call on Hold

G.729/G.711

Agent initiated transfer

3

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Transfer to CVP Complete (Call in Queue)

SIP

6

5

Contact Center

VXMLGateway

4

VXMLService Control

SIPRTP

JTAPI

Call Control


G.729/G.711

G.729/G.711

CUBE 78

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SIP

Contact Center

VXML

Gateway

1VXML

Service Control

SIP

RTPJTAPI

Call Control


G.711

G729 provided SPsupports mid-callcodec renegotiation(likely & preferred)

G711 if SP does notsupport mid-callcodec renegotiationrequires DSP s onCUBE

G.729

CUBE

Transfer to CVP Mixed Codec

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SIP

2

Contact Center

VXMLGateway

1VXMLService Control

SIPRTP

JTAPI

Call Control


G.729

Call on Hold

G.711

G729 (or G711if SP doesnt

support midcallrenegotiation)

**Note mixed codecsrequired for agent

Agent-VXML re-queueusing G711 becauseprompts are G711 forspeech portion

CUBE

3

Agent Warm Transfer to Queue

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SIP

Contact Center

VXML

Gateway

4

VXMLService Control

SIPRTP

JTAPI

Call Control


G.729

G.711

G729 (or G711if SP doesnt

support midcallrenegotiation)

**CUCM disengagesDSPs while call in

queue

DSPs required for transcoding

CUBE

Agent Warm Transfer to Queue

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SIP

6

9

75

Contact Center

VXML

Gateway

CUBE 8

VXMLService Control

SIP

RTPJTAPI

Call Control

CUSP


G.729/G.711

G.729/G.711

Agent Warm Transfer to Queue (Complete)

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Best Practices

Use flat codecs if possible to minimize complex DSP sizing G711Preferred

If mixed codecs are used, detailed call flows will be required tounderstand DSP needs. Most common driver for mixed codecs iswhen 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 (maydrive the need for more DSPs in some call flows)

Use single encoding for CVP prompts (G711), even in mixed codecenvironments

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Best Practices - Continued

Understand SP/carrier capabilities for TBNT, especially howREFERs are handled from the enterprise back into the carriernetwork

Do careful capacity engineering of all the components (i.e.conferencing, transfers & DSPs)

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Customer Case Study

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Global enterprise software company

North America serves 3.1 million SMEs, healthcare andspecialized 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 softwareproducts

Customer Case Study

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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 werestopped in working hours)

Fax would recommend as a standalone project (carrier did not support T.38)

Outbound calls required authentication with owed number

Customer Case Study continued

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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

Customer Case Study continued

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Migration Strategy

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Migration from Legacy Platforms

H.323 (Gatekeeper) to SIP Migration

Hosted IVR and Pre-Routing SIP Migration

Best Practices & Recommendations

Migration Strategy

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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 hair-pinning calls)

Velocity of features (i.e. Service Provider & Enterprisemid-call codec renegotiation, Courtesy Call Back, etc.)

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

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

Begin with End in Mind

Understanding desired End-State Architecture Developing concept of phased migration to End-State

Leverage existing Assets (ICM is foundation)

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

SIP Chat

Email

Web Portal

Video Content

Phone

Kiosk

Migration from Legacy Platforms

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H323 (Gatekeeper) to SIP Migration

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Carrier

VoIP/DATA WAN

Private LinkICMICM

ICMICM

Carrier

Agents

Phones

Agents

Phones

Agents

Phones

Agents

Phones

Contact Centers Contact Centers

Queuing Locally

Gatekeeper Gatekeeper

PG

PG PG

PG

Data Center A Data Center B

Queuing Locally

H.323 Gatekeeper Routing

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Carrier

VoIP/DATA WAN

Private LinkICMICM

ICMICM

Carrier

Agents

Phones

Agents

Phones

Agents

Phones

Agents

Phones


Gatekeeper Gatekeeper

PG

PG PG

PG


H.323 Gatekeeper Routing Continued

ICMICMICMICMPG PG

CUSP adds dial plancentralization

Centralized Queueing

H323

Component

Removalincluding

RAS Dial

Peers onGateways

CUBECUBE

BRKCCT-3007: Upgrade Planning forCisco Unified Intelligent ContactManagement and Cisco UnifiedCustomer Voice Portal

BRKUCC-1021: Cisco UnifiedCustomer Voice Portal and VideoContact Center Update and Roadmap

BRKUCC-2020: Planning andDesigning a Cisco Unified Customer

Voice Portal Deployment

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TDM

G7XX

Media (RTP)

SIP Call Control

RAS/GKTMP

Gatekeeper

TDM

ICM

H.323 to SIP Migration

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G7XX


TDM

G7XX

CUBE

Gatekeeper

G7XX

TDM

VXMLGateway

Hybrid

Environment

ICM & UCCE

Scripts

H.323 to SIP Migration Continued

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G7XX


TDM

G7XX G7XX

CUBE

VXML

Gateway

Hybrid

Environment

ICM & UCCE

Scripts

H.323 to SIP Migration Continued

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Hosted IVR & Pre-Routing SIPMigration

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Carrier

VoIP/DATA WAN

Private LinkICMICM

ICMICM

Carrier

Agents

Phones

Agents

Phones

Agents

Phones

Agents

Phones


Queuing Locally

PG

PG PG

PG


Queuing Locally

Hosted IVR and Pre-Routing

NIC-A NIC-B

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Carrier

VoIP/DATA WAN

Private LinkICMICM

ICMICM

Carrier

Agents

Phones

Agents

Phones

Agents

Phones

Agents

Phones


Queuing Locally

PG

PG PG

PG


Queuing Locally

Hosted IVR and Pre-Routing Continued

NIC-A NIC-B

PG PG

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Carrier

VoIP/DATA WAN

Private LinkICMICM

ICMICM

Carrier

Agents

Phones

Agents

Phones

Agents

Phones

Agents

Phones


PG

PG PG

PG


Hosted IVR and Pre-Routing Continued

PG PG

Centralized Queuing

CUBE CUBE

UCCE

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TDM

TDM

TDM

TBNT

Hosted IVR

ICM

Hosted IVR & NIC SIP Migration

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G7XX


TDM

TDM

TDM

TBNT

Hosted IVR CUBE

VXML

Gateway

Hosted IVR & NIC SIP Migration Continued

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G7XX


CUBE

VXML

Gateway

Hosted IVR & NIC SIP Migration Continued

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IP Carrier

VoIP/DATA WAN

Private Link

ICMICM ICMICM

IP Carrier

Agents

Phones

Agents

Phones

Agents

Phones

Agents

Phones


PG

PG PG

PG


3rd Party SBC to CUBE

PG PG

SBC SBC

Enterprise

SBC

CUBE CUBECUBE CUBECUBE

ContactCenter SBC

Non-contactCenter SIP Traffic

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

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-IParchitectures 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 invocationand/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 yourbusiness if they happen all at once, or in sequence

Best Practices & Recommendations

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Architectural End state

Phase 0

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Carrier TDM

IP WAN

Private Link

Outsource Providers

Enabled via Corporate ICM

ICMICM ICMICM

Carrier TDM

Outsource Providers

Enabled via Corporate UCCE


Agents

FinesseCAD/CTIOS

Contact Centers

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

PG PG

GatekeeperGatekeeper

TB&T through

TDM cloud

Architectural Migration

Phase 1

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Carrier TDM

IP WAN

Private Link

Outsource Providers


Enterprise

Queue & IVR

ICMICM ICMICM

Carrier TDM

Outsource Providers



Agents

CVPCVPCVP

Email

& Web

Interaction CVPCVPCVP

Email

& Web

Interaction

FinesseCAD/CTIOS

Contact Centers

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

PG PGCUPS

GatekeeperGatekeeper

CUPS

Architectural Migration Continued

Phase 2

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Carrier IP

IP WAN

Private Link

Outsource Providers


Enterprise

Queue & IVR

ICMICM ICMICM

Carrier IP

Outsource Providers



Agents

CUSP CUSP

CVPCVPCVP

Email

& Web

Interaction CVPCVPCVP

Email

& Web

InteractionOutbound SIP Dialer Outbound SIP Dialer

FinesseCAD/CTIOS

Contact Centers

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Agents

Cisco or non-Cisco

Phones

Social Media(Standalone)

Social Media(Standalone)

PG PG

CUBE CUBERecording

(Media Sense)

Recording

(Media Sense)


Phase 3A

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SIP

IP Access

PSTN/Internet

Access

SIP

HTTP

Services Advertisement Framework (SAF)


Contacts Contacts

Recording

(Media Sense) XML Routing

Social Media(Blogging)

Presence

XMPP (Jabber)

SME

VideoSIP HTTP

User3rd PartyLTRUCC-2150: CiscoUnified CommunicationsSIP Trunking, SessionManagement, and ServiceAdvertisement Framework


Phase 3B

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IP Access

PSTN/Internet

Access

SIP



Contacts Contacts

User3rd Party


Shared Applications

Recording(Media Sense)

XML Routing

Social Miner(Blogging)

HTTP

SMEHTTP

SIP

PresenceXMPP (Jabber)

Video

SIP

SME

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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

Security BRKUCC-2002: Securing Enterprise Voice Network and Applications

BRKUCC-3501: CUCM Security

Miscellaneous 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

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Cisco.com SIP Trunk and CUBE Resources

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

Cisco CommunicationsTransformations 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_voice_portal/srnd/7x/cvp_gtwy.html#wp1048342
http://www.cisco.com/go/cubehttp://www.cisco.com/go/cubehttp://www.cisco.com/go/interoperabilityhttp://www.cisco.com/go/interoperabilityhttp://www.cisco.com/go/srndhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.htmlhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.htmlhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/trunks.htmlhttp://www.cisco.com/go/srndhttp://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/customer_voice_portal/srnd/7x/cvp_gtwy.htmlhttp://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/customer_voice_portal/srnd/7x/cvp_gtwy.htmlhttp://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/customer_voice_portal/srnd/7x/cvp_gtwy.htmlhttp://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/customer_voice_portal/srnd/7x/cvp_gtwy.htmlhttp://www.cisco.com/go/srndhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/trunks.htmlhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.htmlhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.htmlhttp://www.cisco.com/go/srndhttp://www.cisco.com/go/interoperabilityhttp://www.cisco.com/go/interoperabilityhttp://www.cisco.com/go/cubehttp://www.cisco.com/go/cube

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Recommended Reading

Continue your Networkers at Cisco Livelearning experience with further readingfrom Cisco Press

Check the Recommended Reading flyerfor 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: RueGreen, scheduled to be published late2011

Available Onsite at the

Cisco Company Store

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Q&A

Complete Your Online

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Receive 25 Cisco Preferred Access points for each sessionevaluation you complete.

Give us your feedback and you could win fabulous prizes. Points arecalculated on a daily basis. Winners will be notified by email afterJuly 22nd.

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

Dont forget to activate your Cisco Live and Networkers Virtualaccount for access to all session materials, communities, and on-

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Complete Your OnlineSession Evaluation
http://www.ciscolivevirtual.com/http://www.ciscolivevirtual.com/

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Visit the Cisco Store forRelated Titles

http://theciscostores.com
http://theciscostore.com/http://theciscostore.com/

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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 SIP SIP SIP

H.323 SIP

H.245-Alphanumeric

NOTIFY

H.245-Signal NOTIFY

RFC2833 NOTIFY

H.245-Alphanumeric

RFC2833

H.245-Signal RFC2833

RFC2833 RFC2833

H.245-alphanumeric

KPML

H.245-Signal KPML

Voice In-Band* RFC2833

H.323 H.323

H.245-Alphanumeric

H.245-Alphanumeric

H.245-Signal H.245-Signal

RFC2833 RFC2833

H.245-Alphanumeric

RFC2833

H.245-Signal RFC2833

Voice In-Band* RFC2833

SIP SIP

NOTIFY NOTIFY

RFC2833 NOTIFY

RFC2833 RFC2833

KPML KPML

Voice In-Band*

RFC2833

All DTMF InterworkingIs Bidirectional

*Requires Transcoder DSP

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Call Admissions Control

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

High Water Mark

Low Water MarkTotal Calls, CPU,

MemoryCUBE

Call #1

Call #2

Call #3 Rejected by CUBE

dial-peer voice 1 voip

max-conn 2

call spike call-number [steps number-of-steps size milliseconds]

call spike 10 steps 5 size 200

Call SpikeDetection

Max Calls perDestination

CUBECall #3

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

CUBE

If a call spike is detected, reject calls

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SIP Trunk Best Practices

Use CUBE as the onsite enterprise

Border Element to SIP DO-EO conversion

Normalize traffic SP UNI

Interconnect/share a SIP trunk todifferent 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

The SIP trunk market is maturing

Plan and execute thoroughtesting before production

Most interop issues can beresolved with targetedconfiguration changes andprotocol normalization

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 andConfiguration App Notes

Basic B2BUA Call Flow

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Major difference

between B2BUA

and Proxy. Proxycantdo this

INVITE

200ACK

INVITE

200

ACK

BYE

BYE

200

BYE

200

SIP

UA-1

SIP

UA-2B2BUA

Basic B2BUA Call FlowWhy CVP and UCM (and others) are a B2BUA

200

INVITE

SIP

UA-3And here aswell

Basic SIP Call Flow

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Record Route

is disabled for

faster failover

Invite

GED-125 new call

Send to VRU to Queue callusing VXML GW label

IP PHONE

CVP

Session Controller VXML GWCVP ICM Egress VGW/

UC ManagerSIP

Proxy

Invite

GED-125

IOS VBexecutes vxml

scripts

2-way RTP

200 OK

200 OK

Do we need to

describe this better?Connect/

Answer

HTTP vxml

HTTP New Call

Invite

Invite

200 OK

200 OK

VRU leg request instruction.

Micro app plays prompts &collects digits

alerting

play script

Invite

Basic SIP Call Flow (part 2)

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Record

Route is

disabled forfaster

failover


GED-125 script resultsConnect / Transfer toagent or queuing then

connect to agent

IP PHONE

CVP

Session Controller VXML GWCVP ICM Egress VGW/

UC ManagerSIP

ProxyFlow-Through RTP

200 OK

200 OK

Invite

Caller O

HTTP result

agent label

Invite

BYE200 OK

Ring MGWIn 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 agenttransfer completed

Re-Invite

200 OK

Tclplays

ringing

tone

RTP

ACK

ACK

Only a B2BUAcan do this


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Record

Route isdisabled for

faster failover

as c S Ca o (pa t 3)

IP PHONE

CVP

Session Controller VXML GWCVP ICM Either Egress

VGW or UC

Manager

SIP

ProxyFlow-Through 2-way RTP

ACK

200 OK

200 OK

SIP/SCCP

BYE

200 OK

Re-Invite

200 OK

Invite Ring MGW

ACK

Flow-Through 2-way RTP

Answer

Invite

Release Port IP Transfer (SIP Refer)

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Record

Route isdisabled for

faster

failover

( )

GED-125 script resultsConnect / Transfer toagent using Refer Label

IP PHONE

CVP

VXML GWCVP ICM Egress VGW/

UC ManagerSIP

Proxy

Flow-Through RTP

Caller OHTTP result

Refer Label Z

BYE

200 OK

Refer (agent label Z)

202 Accepted

Notify ringing

180 ringing

InviteInvite

180 ringing

200 OK

SIP/SCCP

answer

200 OK

Notify 200

BYE

CVP will beout of the loop

CVP is out of the call,port license is released

CVP

Session Controller

Release Port 302 Transfer Immediate

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Record

Route is offfor faster

failover

GED-125 script resultsConnect / Transfer toagent using Refer Label

IP PHONE

CVP


UC ManagerSIP

Proxy

Refer Label Z

Notify ringing

180 ringing

Invite

Invite Z

ACK

180 ringing

200 OK

SIP/SCCP

answer200 OK

Notify 200 OK

BYE

CVP is out of the call,port license is released

Invite (SDP-a)

Dialog is notestablished, use 302

Redirect; not Refer

302 (label)

302 (label)

ACK

Session Controller

InviteInvite

Release Trunk TDM Transfer (SIP Info)

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RecordRoute is off

for faster

failover

Release Trunk TDM Transfer (SIP Info)

GED-125 script resultsConnect / Transfer to agent usingRefer Label replacement to DTMF

*8 Label and ICM translation route to

preserve call context. ECC UUI isoptionally set for outbound delivery

IP PHONE

CVP


UC ManagerSIP

ProxyFlow Through RTP

Caller OHTTP result

**Refer Label**

BYE

200 OK

BYE (GTDs UUS)

200 OK

CVP is outof the call

Session Controller

RTP NTEBased on

RFC 2833

NTE (DTMF)

digits are

inband

BYE)

200 OK

Date post:	04-Apr-2018
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