CUBE-EnT Preso For HCS Deployathon - V5

CUBE(ENT) as Replacement for
CUBE(SP) & MetaSwitch Perimeta

in HCS Solution
John Vickroy - Product Line Manager
j_vickroy@cisco.com
May 2016
2010 Cisco and/or its affiliates. All rights reserved.
Cisco Confidential
Disclaimer
The Cisco products, service or features identified in this document may not yet be
available or may not be available in all areas and may be subject to change without
notice. Consult your local Cisco business contact for information on the products or
services available in your area. You can find additional information via Ciscos World
Wide Web server at http://www.cisco.com. Actual performance and environmental costs
of Cisco products will vary depending on individual customer configurations and
conditions.
Discussion Agenda
Market Overview & CUBE Market Positioning
CUBE-ENT relevance to Service Providers
Introducing mVRF and mTenancy on CUBE-ENT
CUBE-ENT features relevant to HCS
CUBE-ENT features compared to CUBE-SP
HCS Solution Architecture with CUBE-ENT
CUBE-ENT as Part of Cisco Call Center Solution (OPTIONAL)
CUBE-ENT voice security policy solutions (OPTIONAL)
CUBE-SP to CUBE-ENT license migration.
SIP Trunk Market Trends
NORTH AMERICA: Per FCC, TDM to SIP trunk

transition in U.S. is about 36% complete thru
FY15..
SIP Session Deployment Growth
Up from 30% at end of FY14.
Accelerating going forward into CY16 thru CY19.
EUROPE: Huge transition is now underway in

Europe with ILECs in most countries.
Approximately 10%-12% transition from ISDN to
SIP trunks
DT, BT, Swisscom announced termination of ISDN
by 2019.
AJPC: Asia, except Australia and Japan, are at

very early phase
Total Market: Worldwide, less than 10% of SIP trunk

transition has occurred so far.
SPs are (desperately) looking for ways to

differentiate SIP trunk services.
Cumulative Total: ~65 M sessions
Remaining Market Transition: ~630 M sessions
Telco Customer Strategies

for VOIP Service Deployment
66%
Cisco Unified Border Element (CUBE)

Overview of Ciscos General Purpose SBC
Enterprise 1
IP PSTN
CUBE
IP
SIP
Enterprise 2
CUBE
IP
SIP
Rich Media (Voice, Video Content) Rich Media
Session
Control
Call Admissions
Control
Trunk Routing
Ensuring QoS
Statistics and Billing
Redundancy (HA)
Scalability
Security
Interworking
Demarcation
Encryption
Authentication
Registration
SIP Protection
Voice Policy
Firewall Placement
Toll Fraud
SIP - SIP
H.323 - SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fault Isolation
Topology Hiding
Network Borders
L5/L7 Protocol
Demarcation
CUBE INTEROPERABILITY
Verified by
Proven Interoperability and

Interworking with
Service Providers Worldwide
Validated with Service

Providers World-Wide
Independently Tested
with 3-Party PBXs in
TekVizion Labs
Standards based SIP
protocols
Cisco Interoperability Portal:

www.cisco.com/go/interoperability
CUBE & Cisco UCM Session Management Edition (SME)

Simplify your infrastructure
CUBE provides session border control
between IP networks
Demarcation
SIP TRUNK TO CUBE
Interworking
Session control
CUBE
Mobile
Cisco B2B
Security
Cisco SME centralizes
network control
Cisco Session
Management
IM, Presence,
Voicemail
Video
Centralizes dial plan

Centralized applications
Aggregates PBXs
3rd Party IP
PBX
TDM PBX
CUBE: Primary Strategic Differentiators

Unmatched by competitors SBCs
SBC Integration
on the Router
Leverages
installed base
and knowledge
base
Integrated SBC
and TDM
Gateway
Simplifies
transition from
to IP PSTN
Broadest Scale of
price
performance
Voice
Security Policy
Solutions for the

smallest to
largest
deployments
TDOS
protection with
granular policybased
enforcement
Enables flexible deployment models:

centralized, distributed, hybrid
Integration with
Cisco
Collaboration
Cisco Unified
CM recording
solutions
CVP & UCCX
call center
solutions
WEBEX
integration for
Cloud Connect
Audio
Scalability
Unequaled scalability in price and performance
ASR 1006
& CUSP
ASR 1004
ASR 1006
50150
2035
Virtualized CUBE (with CUSP load balancing)

scales across capacity spectrum
4400 ISR
Calls per
Second
17
3900 ISR
812
4300 ISR
NanoCUBE
2900 ISR
<5
Capacity: 3,0006,000
ASR 1001-X
ASR 1002-X
Capacity: 16,000
Capacity:
10,000 14,000
ASR +
virtualized CUSP
Capacity: 8002,500
64,000 sessions enabled by

CUBE clustering
Capacity: 1001000
800 ISR Capacity: 100600
Capacity: Up
to 50
<50
500600
1,000
2,500
6,000
Active Voice Call (Session) Capacity
12,000
16,000
64,000
CUBE-Enterprise Value Add Use Cases - Examples

Network-based
Media Recording
(voice or video)
NICE
MediaSense
RTP
SIP
SIP
RTP
SIP
SP SIP
SBC
CUBE
CVP
IVR
IVR & CPA

Integration for
Call Center
SIP
SBC
CUBE
ASR
Server
INSPECT - SCORE
Voice Security
Policy for TDOS
Mitigation
TEST
Inbound
Calls
A
No Cost
Moderate- High Cost
ISR-4K
Migration from
TDM Circuits to
SIP Trunks
PROBE
SP IP
Network
High Cost
TDM
Traditional
PSTN
SIP
H.323
CUBE
SIP
SBC
SP VOIP
Servivces
Enterprise SBC Market

Infonetics Estimates of Market Share Trends
Infonetics Research
Market Analysis CY2014 Revenue
26%
35%
14%
13%
4Q15
Fcst
Note: There are at least 10 other Vendors not shown in
this Chart that would comprise OTHER
11%
Cisco
Oracle / Acme
Audio Codes
Other
Sonus
CUBE: Enterprise SBC Market Leadership
SP transition to SIP services worldwide is creating rapidly growing demand for SBC
CUBE Enterprise Intellectual Property is 100% owned by Cisco.
CUBE is the Enterprise SBC leader (ranked #1 in global market share by INFONETICS/IHS from CY2013 thru CY2015)
CUBE is Used by 17,000+ organizations with 14 million+ trunk sessions in 160+ countries
CUBE offers outstanding interoperability with both SP SIP services and 3rd party IP-PBXs
CUBE Primary Differentiators:
Router Integration (see chart)
Price / Performance Scalability
TDM/VOIP integration
CUBE adds value to Cisco UC Solutions:
Call Center
Voice Security Policy
Call Recording
WEBEX CCA
HCS
Major CUBE Customers:
JP Morgan Chase
Royal Bank of Scotland
Chevron
Walmart
GM
FORD
GEICO
Apple
50150
2035
17
4400 ISR
3900 ISR
Capacity:
3,000-6,000
4300 ISR
812 NanoCUBE
2900 ISR
<5 800 ISR

Capacity
: Up to
50
ASR 1001-X
ASR 1002-X
Virtualized CUBE (with CUSP load

balancing) scales across capacity
spectrum
Calls per
Second
ASR 1004
ASR 1006
Capacity:
16,000
Capacity:
10,000
14,000
Capacity:
800-2,500
Capacity:
100-1000
Capacity:
100-600
<50
500600 1,000
2,500
6,000
12,000
16,000
CUBE Enterprise Training Labs
CUBE Enterprise On-Line Training:
https://cisco.box.com/SIPLabTheoryandAccess
CUBE Technical Resources:
https://cisco.box.com/CUBE
SIP Services Operational Vectors faced by

Service Providers
Managed
Services
MPLS
Audio/
Video
SMB
Large
Enterprise
Internet
Audio
Only
Hosted
Services
CUBE Strategic Value to SPs for SIP Services:

Lowest Possible Operation Costs
Managed Services
SIP TRUNK
Call control on Customer Prem
MPLS
Mid-range
CUBE
SMB
Nano-CUBE
with 3PCC
CUBE
WEBEX CCA-SP
Internet
Multi-Tenant CUBE
for HCS
Hosted Services
SIP Line-side
Call control in the cloud
Large
Enterprise
CUBE
Large
Enterprise
Efficient Network Architecture for SIP Services

Separation of SBC function provides lowest Operational Costs
IP PSTN
Aggregation SBC
Multi Tennant
High Session Capacity
Normalization
Policy
Services
SBC

Aggregation SBC
Multi Tennant
IP PSTN
SBC
Customer Prem SBC

(Single Tenant CUBE-Ent)
Normalization
Policy
Services (recording)
Reduces risk of Aggregation SBC Failure from unusual SIP normalization

Allows greater utilization of Aggregation SBC due to no processing of SIP normalization
Ensures Uniformity of SIP interaction from Customer Prem to Aggregation SBC
Enables use of provisioning templates for customers with the same IP PBX

IP PSTN
Aggregation SBC
Multi Tennant
HCS SBC
mTENANT CUBE-Ent
SBC
Normalization
Policy
Call Center
Customer Prem SBC

Single Tenant CUBE-Ent
Normalization
Policy
CUBE
CUBE

IP PSTN
Aggregation SBC
Multi Tennant
HCS SBC
mTENANT CUBE-Ent
SBC
Normalization
Policy
Call Center
Customer Prem SBC

Single Tenant CUBE-Ent
Normalization
Policy
CUBE
CUBE
IP PSTN
MPLS VPN
CUBE
CUBE as Part of CPE Strategy for Broadsoft

A complete CPE Solution for Hosted Services
SP Benefit w/ CUBE CPE
nanoCUBE Features for 3PCC
CPE for both Hosted and Lineside

Services.
Reduce Costs of Operation
Expand SIP-based services offering
Standards-based SIP proxy endpoint

registration
Local Survivability
PSTN Survivability with FXO
LAN & WAN VQM
Call Pre-emption (eg 911)
SBC
nanoCUBE
Platforms
Business
Internet
C88X
C89X
SPIAD29XX
SIP trunk
Connection
MPLS VPN
LAN-I
SIP Lineside
Connection
Certified demarcation
CUBE
CUBE
Current nanoCUBE
3PCC Partners
Swisscom
Windstream
CLARO
IP-PBX
IP-Phone
LINESIDE SERVICE
Competitors
ORACLE
Edgewater
Adtran
AudioCodes
CUCM 10.0 Network Based Recording Requires CUBE

Delivers Industrys Most Flexible Call Recording Architecture
SP IP Network
CUBE
BENEFITS OF CUCM NBR with

CUBE:
Enhanced Control CUCM has policy

control over media forking on CUBE &
GWs.
Better Bandwidth Utilization Any
Distributed
CUCM can invoke media forking on any
Recording
CUCM
CUBE
Centralized
Flexibility distributed or centralized
Recording
recording architecture, uses any 3rd
party recording server
Enterprise Network
Mobility - Allows recording of any
Cisco mobile client (i.e. DVO, SNR,
Via API, CUCM controls CUBE to
Requires Upgrade
Jabber)
perform Media Forking for any Call Session
To
CUCM 10.X
Improved
Compliance
- Ensures
regulatory compliance for mandatory
recording
CUBE
Use Case: SIP Trunking to Cisco WebEx

Cost-effective audio conferencing for your WebEx web conferences
TDM PSTN
Requirements
WEBEX
Replaces TDM audio connection to WebEx with VoIP

using SIP signaling.
WebEx cloud becomes a portal off of enterprise WAN.
CUBE
How
A
Enterprise
IP WAN
(MPLS)
CUBE
CUBE reduces SIP protocol chatter between IP-PBX

and WebEx cloud through SIP normalization.
CUBE enables SIP sessions from ALL enterprise sites to

WebEx to avoid hairpin media flows.
CUBE provides high performance for signaling and media

transport of WebEx.
Headquarters
Benefit
Branch
Office
Branch
Office
Branch
Office
Dramatic savings thru elimination of TDM, plus excellent

conference experience thru efficient network usage.
CUBE Architecture Flexibility

Efficiently Support All SIP Architectures for Voice or Video Services
Centralized
Distributed
IP PSTN
IP PSTN
Enterprise
IP WAN
Enterprise
IP WAN
CUBE
CUBE
Hybrid
CUBE
CUBE
CUBE
IP PSTN
Enterprise
IP WAN
CUBE
CUBE
CUBE
CUBE
CUBE
CUBE
CUBE Deployment Adapts with Collaboration Needs

Multi-party conferencing and Video SP services will be the drivers.
SP
SP
SP
Centralized
Centralized
Distributed
Distributed
Hybrid
Audio Only
1:1
Best
Good
Better
Audio Only
Multiparty
Good
Better
Better
Audio & Video

1:1
Good
Better
Better
Audio & Video

Multiparty
Worst
Best
Better
Audio, Video, Desktop

Share 1:1 or multi-party
Worst
Best
Better
Collaboration Service
Hybrid
Use Case: SIP Trunking to Cisco WebEx

Cost-effective audio conferencing for your WebEx web conferences
Requirements
TDM PSTN
Replaces TDM audio connection to WebEx with VoIP

using SIP signaling.
WebEx cloud becomes a portal off of enterprise WAN.
WEBEX
CUBE
How
CUBE normalizes SIP protocol interaction between

IP-PBX and WebEx cloud.
CUBE enables SIP sessions from ALL enterprise sites to

WebEx to avoid hairpin media flows.
CUBE provides high performance for signaling and media

transport of WebEx.
A
Enterprise
IP WAN
(MPLS)
CUBE
Headquarters
Benefits
Branch
Office
Branch
Office
Branch
Office
Dramatic savings thru elimination of TDM, plus excellent

conference experience thru efficient network usage.
CUBE-ENT support for video and BFCP will allow

complete consolidation of WEBEX onto SIP.
CCA-SP - Topology
Customer
No Cisco UC
Requirement
SIP Trunk
Cisco WebEx
Collaboration Cloud
SP
SBC
WebEx
CUBE
SP Network
Leverage SP Network
TDM PRIs
SP Partner
Service Wrapper
Service Aggregation
Wholesale models
Cisco
HCS integration
Foundation architecture
for future network enabled
cloud services
WebEx Service
DMARC
Enterprise A
Network
Location 1
E1/T1, E3, T3
Ethernet, DSL
E1/T1, E3, T3
Ethernet, DSL
CUBE
PSTN
CUBE
CUBE
Enterprise B
Network
Location N
On-net Users - Users on Corporate

network
Location 1
Off-net Users Will come

through PSTN
Service Provider Service Level Components

Service Provider
Customer
WebEx
WebEx
(Conferencing
Platform)
On-net
PSTN GW
4
PSTN
Off-net
On Premise Equipment
- IP PBX, SBC, Gateways

- Phones
6
IP/MPLS Access & Network
SIP Trunks
Off-net Access, PSTN, Phone

Numbers
Managed Services
- Service Set-up
- Managed Day 2 Support
WebEx Services
- Data Meetings
- CCA Ports
CCA-SP & HCS Service Integration

Cisco WebEx
Collaboration Cloud
Audio
Bridges
WebEx CCA
Service Demarc
SIP
SIP
(dedicated)
PSTN/SIP
Network
Service Provider Network
Aggregation
Layer
CUBE-ENT
UC
Layer
(DC)
HCS WAN
HCS WAN
Customer A
Customer B
Customer Premise (CPE)
Customer C
Local Breakout
PSTN/SIP
Network
CUBE Multi VRF & Multi Tenancy Support
Introduction
Virtual Route Forwarding (VRF) is a technique which creates multiple virtual
independent networks within router
CUBE Enterprise Integrated with VRF enables a number of new use cases for
CUBE, including:
Logical separation of multiple CUBE call routing domains.
Enables multi-tenant deployments
Enablement of intra and inter VRF routing of voice and video calls
between VRF domains, without sharing IP addressing.
Security can be improved for calls between SP domains
Enablement of IP address Overlap with Multi VRF feature provide
seamless integration of new networks.
Allows greater flexibility in connecting to multiple SPs
Allows multiple legacy IP-PBXs with overlapping DN ranges to utilize
CUBE.
CUBE Support for Multi-VRF

Enables IP routing separation of PSTN or User domains
Enables overlapping IP
addresses between different
session targets (e.g. SPs)
Enables independent SIP
User Agent definitions for
each session target (e.g. SP)
SP1
SP2
SP3
V
R
F
1
V
R
F
2
V
R
F
3
CUBE Support for multi-Tenancy

Enables Dial Plan and IP routing separation of User Domains
Enables multiple independent
User Domains
Allows overlapping DN ranges
betwee User Domains
SP1
SP2
SP3
V
R
F
1
V
R
F
2
A
V
R
F
3
CUBE mVRF/mTenant Flexible Configuration Options

CUBE mVRF
CUBE mTENANCY
Intra VRF call routing
Allowed
Required
Inter VRF call routing
Allowed
Not Allowed
Overlapping Dial Plan
Allowed,
(Requires use of Dial Peer
Group DPG CLI)
Allowed
IP Session Target Overlap
Allowed
Allowed
IP Address Overlap
Allowed
Allowed
Not Allowed
Required
Yes
Yes
Localized User Agent Definition

vCUBE Supported
mVRF Feature Technical Details
Listen sockets on UDP, TCP and TLS transports based on the VRF
Provision to configure RTP port ranges for each VRF and allocation
of Local RTP ports based upon VRF.
Stateful switchover of calls with VRF on both Inbox and Box-to-Box

HA. Check pointing VRFID onto standby router.
Ability to route the VoIP calls across different VRFs without need of
Route Leaks.
Show command outputs enhancement to display the VRF IDs for

voice and video calls.
Overlap IP address support on CUBE using different VRFs
Categories of mVRF Usage
Multi-VRF
Basic Configuration
Inbound Dial Peer Match config
Inter / Intra VRF Call Routing config
Routing w/ Overlapped DN config
Routing w/ Overlapped IP config
Multi-Tenant
SIP UA Feature config

Multi-tenant config
CUBE Multi VRF - Basic Configuration
VRF
1
VRF
2
CUBE
ip vrf vrf1
rd 1:1
interface GigabitEthernet0/0/0
ip address 7.44.44.13 255.255.0.0
ip vrf forwarding vrf1
dial-peer voice 1 voip

voice-class sip bind all interface GigabitEthernet0/0/0
ip vrf vrf2
rd 2:2
ip address 6.44.44.13 255.255.0.0

VRF association by binding dial-peer to appropriate interface

Up to 54 different VRFs supported
CUBE Multi VRF Inter/Intra VRF Routing

VRF
1
VRF
2
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
ip address 7.44.44.13 255.255.0.0

ip address 6.44.44.13 255.255.0.0


incoming called-number 2000


destination-pattern 2000
session-target ipv4: 10.1.1.1
Intra VRF Routing

Regular outbound dial-peer matching logic

May not work with over-lapped DNs

session-target ipv4:10.2.2.2

CUBE Multi VRF Routing w/ Overlapped DN

VRF
1
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
ip address 6.44.44.13 255.255.0.0

ip address 7.44.44.13 255.255.0.0

voice class dpg 222
voice class dpg 111
dial-peer 22 preference 1

destination dpg 111
dial-peer voip 11 voip
VRF
2
Route based on
outbound
dial-peer group

Use outbound dial-peer groups to route calls

Inter or Intra VRF supported

destination dpg 222

CUBE Multi VRF Routing w/ Overlapped Local

VRF
1
ip vrf vrf2
rd 2:2
rd 1:1
Overlap local IP
ip address 7.44.44.13 255.255.0.0

voice class dpg 222
voice class dpg 111

destination dpg 222

destination dpg 111

VRF
2
CUBE
ip vrf vrf1
ip address 7.44.44.13 255.255.0.0

IP
Different remote IP
Local and Remote IP overlap supported across VRF

CUBE Multi VRF Routing w/ Overlapped IP

VRF
1
VRF
2
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
Interface GigabitEthernet0/0.1
encapsulation dot1Q 1 native
ip address 2.44.44.9 255.255.255.0
Overlap local IP
w/ VLAN subint
voice class dpg 222
voice class dpg 111
voice-class sip bind all interface GigabitEthernet0/0.2

destination dpg 222

destination dpg 111

Interface GigabitEthernet0/0.2
encapsulation dot1Q 2
ip address 2.44.44.9 255.255.255.0
Different remote IP

CUBE Multi VRF Routing w/ Overlapped IP

VRF
1
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
ip address 7.44.44.13 255.255.0.0
Overlap local IP
ip address 7.44.44.13 255.255.0.0

voice class dpg 222
voice class dpg 111

destination dpg 222

destination dpg 111

VRF
2
Overlap remote IP

CUBE Multi VRF Inbound dial-peer match

VRF
1
VRF
2
CUBE
ip vrf vrf1
ip vrf vrf2
rd 2:2
rd 1:1
ip address 7.44.44.13 255.255.0.0

ip address 6.44.44.13 255.255.0.0

Available in phase 2
Inbound match based on VRF where SIP INVITE received.
For VRF 1, dial-peer 1 matched. For VRF 2, dial-peer 2.
SIP-UA Feature Overview

Most of configs under sip-ua and sip configs added in voice class tenant <>
Registrar and Credentials cli under tenant using different bind and outbound proxy.
Global OB & Bind
Prior to Multi Tenant
Router#show run | sec sip-ua

registrar 1 ipv4:10.64.86.35:9051 expires 3600
credentials username aaaa password 7 06070E204D realm aaaa.com
credentials username bbbb password 7 110B1B0715 realm bbbb.com
Router#show run | sec voice service voip sip
outbound-proxy ipv4:10.64.86.35:9057
Bind control source-interface GigabitEthernet0/1
E164 - aaaa
Registrar - 1
E164 - bbbb
Registrar - 2
Multi Tenant
Router#show run | sec tenant
Voice class tenant 1
bind control source-interface GigabitEthernet0/0
Voice class tenant 2
credentials username bbbb password 7 110B1B0715 realm bbbb.com
OB-1 & Bind-1
E164 - aaaa
E164 - bbbb
Registrar - 1
OB-2 & Bind-2
Registrar - 2
Configurations
Add voice class tenant <>
Router(config)#voice class tenant 1

Router(config-class)#?
VOICECLASS configuration commands:
aaa
sip-ua AAA related configuration
anat
allow alternative network address types IPv4 and IPv6
asserted-id
Configure SIP UA privacy identity settings
associate
Associate a RCB for outgoing calls
asymmetric
Configure global SIP asymmetric payload support
authenticate
Call authentication policy
bind
SIP bind command
.
.
copy-list
Configure list of entities to be sent to peer leg
credentials
User credentials for registration.
outbound-proxy
Configure an Outbound Proxy Server
.
registrar
SIP Registrar config
registration
SIP Registration Options
video
video related config for sip
warn-header
SIP Warning-Header global config
** (Showing only couple of configs as an example. All the configs under tenant be similar to configs under
sip-ua/voice service voip sip)
SIP User Agent Sample Configurations

Configure voice class tenant
Router#sh run | sec voice class tenant 1
voice class tenant 1
credentials number bbbb username bbbb password 7 110B1B0715 realm bbbb.com
bind media source-interface GigabitEthernet0/0
copy-list 1
early-offer forced
Router#
Apply tenant to dialpeer
Router#sh run | sec dial-peer voice 1 voip

session protocol sipv2
session target ipv4:10.64.86.35:9051
session transport udp
voice-class sip tenant 1
Router#
CUBE Multi Tenant Configuration

VRF
1
ip vrf vrf1
rd 1:1
ip address 7.44.44.13 255.255.0.0

registrar ipv4:10.1.1.5 expires 3600
credentials username vrf1 password 7 104F081804 realm vrf1.com
max-forwards 57
retry invite 7
timers trying 100
bind all source-interface GigabitEthernet0/0/0

voice class sip tenant 1

VRF
2
CUBE
ip vrf vrf2
rd 2:2
ip address 6.44.44.13 255.255.0.0

registrar ipv4:10.2.2.5 expires 3600
credentials username vrf1 password 7 104F081804 realm vrf2.com
max-forwards 58
retry invite 5
timers trying 200
bind all source-interface GigabitEthernet0/0/1

voice class sip tenant 2

Key CUBE-ENT Features

Relevant to HCS
November-2015
CUBE-ENT Support for HCS requirement

Requirement
Supported
Comments
Target Date / Plan
MultiVRF / Multi-Tenant
support
Yes
Configured with Dial Peers
ISR-G2 now / ASR1K in July
RFC2833 to NOTIFY dtmf

interworking
No
Only notify to rfc2833 dtmf is supported
Required for offering Dial via Office or

CVP
Delay Offer to Early Offer
Yes
Only for voice, not video
NOW
PRACK interworking for

early media
Yes*
Need to check with HCS on some detail

use case
Probably not used by HCS
Suppression of UPDATE
Yes
Redundant peer and load

balancing
Yes*
Some case might be a gap i.e, 401/407
POC testing would find gaps
Routing for E911 Calls
Yes
Configured with Dialpeers
NOW
BFCP & Video SDP Mlines
Yes
NOW
Setting of DSCP
precedence
Yes
Configured with Dialpeers precedence
NOW
Fixed BlackList
Yes
Configured at L3 with ACLs or at L7 with

Dial Peer File (see example)
NOW
NOW, Used for Webex Interworking
CUBE-ENT support for HCS requirements

Requirement
Supported
Comments
Target Date / Plan
SIP Normalization
Yes
Both inbound and outbound dial peer logic
Now
3rd Party registration
Yes
Req-uri Parameter handlng
Yes
SIP to SIP video
Yes
SIP SIMPLE IM Message

Passing
No
Not on CUBE(SP) either
Roadmap for CY2017
InterVRF Routing
Yes
ISR-G2 now / ASR1K in July
CUBE Based Routing (ie

dialplan on CUBE)
Yes
Supported with external files that map

E164 numbers to VRF, allows HCS
deployment at much lower cost
July 2014
Media Forking / Recording
Yes
Allows new HCS offering for outsourced

recording
July 2014
Dynamic Blacklist
Yes
With Voice Policy Solution
Now
Route Header passing

between IMS and CUCM
??
Under investigation
TBD
ASR, CSR & ISR-G2/4K Feature Comparison

General Platform Features
ASR1K
ISR-G2
4300/4400 (XE3.13.1)
vCUBE (XE3.15+)
High Availability Implementation
Redundancy-Group
Infrastructure
HSRP Based
Redundancy-Group
Infrastructure
Redundancy-Group
Infrastructure
TDM Trunk Failover/Coexistence
Not Available
Exists
Exists
Not Available
Media Forking
XE3.8
15.2.1T
XE3.10
Exists
Software MTP registered to

CUCM (Including HA Support)
XE3.6
Exists
Exists
Exists
DSP Card
SPA-DSP
PVDM2/PVDM3
PVDM4
Not Available
Transcoder registered to CUCM
Not Available
Exists via SCCP
Exists via SCCP (XE3.11)
Not Available
Transcoder Implementation
Local Transcoder Interface

(LTI)
SCCP or
LTI (starting IOS 15.2.3T)
SCCP and LTI
SCCP based on a separate

platform, CUCM controlled
Embedded Packet Capture
Exists
Exists
Exists
Exists
Web-based UC API
XE3.8
15.2.2T
Exists
Exists
Noise Reduction & ASP
Exists
15.2.3T
Exists
Not Available
Call Progress Analysis
XE3.9
15.3.2T
Exists
Not Available
CME/SRST feature set
Not Available
Exists
XE3.11
Not Available
SRTP-RTP Call flows
Exists (NO DSPs needed)
Exists (DSPs required)
Exists (NO DSPs needed)
Exists (No DSPs needed)
VXML GW
Not Available
Exists
Not Available
Not Available
CUBE ENT on ASR1K

Session Capacity with Additional Features: XE 3.13.1 and later
CUBE + Xcoding1
Calls Per
Second
(CPS)
CUBE
FlowThru
Calls
CUBE +
SW MTP
ASR1001-X
50
12000
ASR1002-X2
55
ASR1004 RP2
ASR1006 RP2
Platform
SIP TLS
w/SRTPRTP3
CUBE
Controlled
Recording
CUCM 10.X
Controlled
Recording
Xcoded
Calls
Simultaneous NonXcoded Calls
5000
357
11643
6600
6000
6000
14000
5000
1071
12929
6600
7000
7000
70
16000
8000
2499
13501
10800
8000
8000
70
16000
8000
3927
12073
10800
8000
8000
Refer to this iWE post on Contact Center sizing
For Contact Center deployments, the max recommended capacity for CUBE ENT on an ASR1006 RP2 is no more than 4000
sessions for complex call flows based on 1000 SIP messages per second handled by the platform
Transcoding
Please reach
to on
theG711-G729r8
cs-cube@cisco.com
for capacity
planning
questions
prior to
customer
quotations indicate the
1.
is out
based
(High Complexity
Codec)
sessions.
The numbers
listed
for CUBE+Xcoding
number of transcoded sessions supported based on max DSP capacity as well as non-transcoded sessions. The total of both
should not exceed the CUBE Flow-Thru Calls for the overall
number
sessions
supported
mix
of some transcoded
and
CPS and
Sessionofcounts
listed
in (1) and in
(2)aare
independently
tested. Session
some non-transcoded sessions.
capacity (2) can be achieved at about half the CPS(1) listed here
2. ASR1002-X based on ESP10
ATT
Sprint
CUBE CLUSTERING
Scaling CUBE using CUSP for Load Balancing
Expanding CUBE Capacity with CUSP

CUSP CPS
Ratings
CUBE
ASR
1006
CUBE
ASR
1001
CUBE
ISR-G2 3945E
CUBE CPS -
Max
150
100
40
CUBE CPS
Typical
50
33
15
200
400
4:1
8:1
6:1
12:1
13:1
26:1
750
1500
15:1
30:1
23:1
46:1
50:1
100:1
CUSP-SRE
CPS RR On
CPS RR Off
CUSP UCS-E
CPS RR On
CPS RR Off
RFC2833 to SIP NOTIFY interworking

For basic RTP to RTP call, notify -> RFC2833 is
supported, but RFC2833 -> to notify is NOT supported

For transcoding and SRTP to RTP call, notify <->
RFC2833 dtmf interworking not required.

RTP to RTP should move towards KMPL to RFC4733
Interworking which is on CUBE(ENT) Radar
CUBE SIP Normalization
Step 5: SIP Normalization

SIP profiles is a mechanism to normalize or customize SIP at the
network border to provide interop between incompatible devices
SIP incompatibilities arise due to:
A device rejecting an unknown header (value or
parameter) instead of ignoring it

A device expecting an optional header
value/parameter or can be implemented in

multiple ways
A device sending a value/parameter that must
be changed or suppressed (normalized)

before it leaves/enters the enterprise to comply
with policies
Variations in the SIP standards of how to
achieve certain functions
With CUBE 10.0.1 SIP Profiles
Add user=phone for INVITEs

Incoming
INVITE
sip:5551000@sip.com:5060
SIP/2.0
Outgoing
CUBE
INVITE
sip:5551000@sip.com:5060
user=phone SIP/2.0
voice class sip-profiles 100

request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
Modify a sip: URI to a tel: URI in INVITEs
Incomi
ng
INVITE
sip:2222000020@9.13.24.6:5060
SIP/2.0
Outgoing
CUBE
INVITE
tel:2222000020
SIP/2.0

request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1"
request INVITE sip-header From modify "<sip:(.*)@.*>" "<tel:\1>"
request INVITE sip-header To modify "<sip:(.*)@.*>" "<tel:\1>"
can be applied to inbound SIP

messages as well
More information at www.cisco.com/go/cube > Configure > Configuration Examples and TechNotes
Normalize Outbound SIP Message (Example 1)

SIP Provider
Requirement
For Call Forward & Transfer scenarios back to PSTN, the

Diversion header should match the registered DID of your network
SIP INVITE that CUBE sends

Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
User-Agent: Cisco-SIPGateway/IOS-15.2.3.T
Diversion: <sip:3000@9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
...
m=audio 6001 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
...
Configure
SIP Profiles
Apply to
Dial-peer or
Globally
SIP INVITE that Service Provider expects

Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
.
User-Agent: Cisco-SIPGateway/IOS-15.2.3.T
.
Diversion: <sip:4085266855@sip.abc.com>;
privacy=off;reason=unconditional;screen=yes
.
m=audio 32278 RTP/AVP 18 8 101
..

request INVITE sip-header Diversion modify sip:(.*>) sip:4085266855@sip.abc.com>
request REINVITE sip-header Diversion modify sip:(.*>)
sip:4085266855@sip.abc.com>
description Incoming/outgoing SP
voice-class sip profiles 400
voice service voip

sip
sip profiles 400
For Your
Reference
Normalize Inbound SIP Message (Example 2)

CUBE
Requirement
SIP Diversion header must include a user portion
SIP INVITE received by CUBE

Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
User-Agent: SP-SBC
Diversion: <sip:9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
...
m=audio 6001 RTP/AVP 0 8 18 101
...
Enable Inbound SIP

Profile feature
SIP INVITE CUBE expects

Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
.
User-Agent: SP-SBC
.
Diversion: <sip:1234@abc.com>;
privacy=off;reason=unconditional;screen=yes
.
m=audio 32278 RTP/AVP 18 8 101
..
voice service voip

sip
sip-profiles inbound
Configure Inbound
SIP Profile to add a
dummy user part

request INVITE sip-header Diversion modify sip: sip:1234@
Apply to Dial-peer
or Globally

description Incoming/outgoing SP
voice-class sip profiles 400 inbound
voice service voip

sip
sip profiles 400 inbound
For Your
Reference
SIP Profile Support for NonStandard Headers
SIP Profile support for Non-Standard Headers

Introducing support for adding/copying/removing/modifying non-
standard SIP headers using SIP profiles

A new 'WORD' option has been added to the SIP Profiles CLI chain to
allow the user to configure any non-standard SIP Header

CUBE(config)#voice class sip-profiles 1
CUBE(config-class)#request INVITE sip-header ?
Accept-Contact SIP header Accept-Contact
.
Via
SIP header Via
WORD
The new WORD

option for specifying
unsupported headers
Any other SIP header name
WWW-Authenticate
SIP header WWW-Authenticate
CUBE(config-class)#request INVITE sip-header WORD ?

ADD addition of the header
COPY
Copy a header
MODIFY Modification of a header
REMOVE Removal of a header
CUBE(config-class)#request INVITE sip-header WORD ADD MyCustomHeader : Hussain Ali
SIP Profile Use Cases
CUBE SIP Profiles Examples

1.
Remove update from allow header: -- USE THIS IF YOU DON'T WANT THE PEER SIP
EQUIPMENT TO NOT SEND A MID-CALL SIP UPDATE TO THE CUBE.
request ANY sip-header Allow-Header modify "UPDATE, " ""
response ANY sip-header Allow-Header modify "UPDATE, " ""
2.
Change the host part of the Remote-Party-ID header to @sp.com instead of @CUBE's-IPaddress: -- USE THIS IF YOU WANT TO CHANGE THE DOMAIN PART OF RPID HEADER TO
REFLECT THE SP's DOMAIN. ESPECIALLY TRUE WITH SP's WHO HAVE BROADSOFT AS
THEIR SIP CALL-AGENT
request ANY sip-header Remote-Party-ID modify @9.13.24.5 @SP.com
response ANY sip-header Remote-Party-ID modify @9.13.24.5 @SP.com

3.
Change sendonly to inactive in reINVITEs, 200 Oks and ACK w/SDPs -- USE THIS IF YOU HAVE ONEWAY OR NO-WAY VOICE PATH ISSUES WITH 3rd PARTY SIP EQUIPMENT, ESPECIALLY WHEN THE
SIP PEER DOES NOT COME OUT OF SENDONLY / RECVONLY / INACTIVE STATE AFTER CALL IS
PUT ON HOLD
request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
response 200 sdp-header Audio-Attribute modify "sendonly" "sendrecv"
Apply this to the voip dial-peer towards the SP using "voice-class sip-profiles 1".
The "inactive" attribute can be in ACK message or in a response. To cover all possibilities, this profile
below will convert any/all inactive's / sendonly's to sendrecv.
request any sdp-header Audio-Attribute modify "inactive" "sendrecv"
request any sdp-header Audio-Attribute modify "sendonly" "sendrecv"
response any sdp-header Audio-Attribute modify "inactive" "sendrecv"
response any sdp-header Audio-Attribute modify "sendonly" "sendrecv"

4.
Remove Referred-By from outgoing REFER message: -- USE THIS IF YOU WANT TO SIMULATE
VARIOUS CONDITIONS WHILE REPRODUCING REFER RELATED CALL FLOW ISSUES
request REFER sip-header Referred-By remove
5.
Remove double c-lines from IOS SDP: -- USE THIS IF YOU ENCOUNTER INTEROP ISSUES
RELATED TO IOS SENDING SESSION AND MEDIA LEVEL c= LINES IN SDP
If having two c-lines in the SDP causes problems you can filter out whatever level c-line.
The excerpt below is for removing the media level c-line:
request ANY sdp-header Audio-Connection-Info remove
response ANY sdp-header Audio-Connection-Info remove
6.
Add user=phone to the P-Asserted-Identity URI: -- USE IF YOU NEED TO ADD user=phone
PARAMETER ADDED TO THE PAID HEADER.
voice class sip-profile 1
request INVITE sip-header P-Asserted-Identity modify "sip:@" "sip:"

7.
8.
Adding Subscription State: Active to NOTIFY messages for MWI: --- USE IF THERE IS AN INTEROP ISSUE
RELATED TO MWI NOTIFY
request NOTIFY sip-header Subscription-State add "Subscription State: Active"
"Subscription State: Active"
<---- add the string

request NOTIFY sip-header Subscription-State modify "" "Subscription State: Active"
nothing add the string "Subscription State: Active
<---- if there is
Add "user=phone" to Req URI -- USE IF SP or PEER WANTS user=phone TO APPEAR IN THE Req URI of SIP
INVITEs
request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"

9.
Convert "sip" to "tel" URI: -- USE IF YOUR SP NEEDS IOS TO SEND tel: URI INSTEAD OF sip: URI
request INVITE sip-header SIP-Req-URI modify "sip .*)@[^ ]+" "tel:\1"
request INVITE sip-header From modify "<sip .)@.>" "<tel:\1>"
request INVITE sip-header To modify "<sip .)@.>" "<tel:\1>
10.
Convert c=0.0.0.0 to c=<valid IP> in SDP of ReINVITEs: -- USE IF YOUR SP (ex. ALCATEL PBX)
DOESN'T SUPPORT c=0.0.0.0 in SDP in SETUPS like CUCM -- CUBE -- ALCATEL PBX
request ReINVITE sdp-header connection-Info modify "0.0.0.0" "10.199.71.100"
request ReINVITE sdp-header Audio-connection-Info modify "0.0.0.0" "10.199.71.100
11.
Add a tag, such as a trunk group indication, in the From header of the INVITE: -- USE THIS IF YOU
WANT TO DISTINGUISH CALLS AS BELONGING TO DIFFERENT TRUNK GROUPS - THE TRUNK
GROUP IS DEFINED BY HITTING A DIAL-PEER WITH A UNIQUE TRUNK-GROUP NUMBER ADDED
BY THE PROFILE ATTACHED TO THAT DIAL-PEER

request INVITE sip-header From MODIFY ">;" ";otg=9523>;"

12.
Disable SIP session timer: -- USE TO AVOID YOUR SP (ex. SONUS PBX) REQUESTING FOR
SESSION TIMER WHEN THE OTHER END (ex. MITEL) DOESN'T SUPPORT SESSION TIMER. MITEL
-- CUBE -- SONUS
request INVITE sip-header Supported modify "timer," ""
request REINVITE sip-header Supported modify "timer," ""
request UPDATE sip-header Supported modify "timer," ""
request INVITE sip-header Min-SE remove
request REINVITE sip-header Min-SE remove
request UPDATE sip-header Min-SE remove
13.
Suppress the advertisement of the INFO message: -- USE IF YOUR SP (e.g. Acme SBC) RESPONDS
WITH AN INFO MESSAGE THAT MAKES THE CALL FLOW FAIL
request ANY sip-header Allow-Header modify "INFO, " ""
response ANY sip-header Allow-Header modify "INFO, " ""
CUBE - Other Useful Call Routing Mechanisms
CUBE Call Routing Features
Understanding Dial-Peer matching

Techniques:
LAN & WAN Dial-Peers
LAN Dial-Peers Dial-peers that are facing towards the IP PBX for sending and receiving calls
to & from the PBX

WAN Dial-Peers Dial-peers that are facing towards the SIP Trunk provider for sending &
receiving calls to & from the provider
Inbound LAN Dial-Peer
Outbound WAN Dial-Peer
Outbound Calls
CUCM SIP Trunk
ITSP SIP Trunk
IP PSTN
CUBE
Inbound Calls
Outbound LAN Dial-Peer
Inbound WAN Dial-Peer
Multiple Destination-Patterns Under Same

Outbound Dial-Peer
Site A
(919)200-2000
Site B
(510)100-1000
Site C
(408)100-1000
G729 Sites
voice class e164-pattern-map 100

e164 919200200.
e164 510100100.
e164 408100100.
destination e164-pattern-map 100
codec g729r8
session target ipv4:10.1.1.1
SIP Trunk
Provides the ability to combine multiple

destination-patterns targeted to the
same destination to be grouped into a
single dial-peer
Up to 5000 entries in a text file

SP SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
G711 Sites

url flash:e164-pattern-map.cfg
destination e164-pattern-map 200
codec g711ulaw
! This is an example of the contents

of E164 patterns text file
stored in flash:e164-patternmap.cfg
9192002010
5101001010
4081001010
Multiple Incoming Patterns Under Same

Incoming Dial-peer
Site A
(919)200-2000
Site B
(510)100-1000
Site C
(408)100-1000
G729 Sites

e164 919200200.
e164 510100100.
e164 408100100.
description Inbound DP via Calling
incoming calling e164-pattern-map 300
codec g729r8
SIP Trunk
Provides the ability to combine multiple

incoming called OR calling numbers on
a single inbound voip dial-peer, reducing
the total number of inbound voip dialpeers required with the same routing
capability
Up to 5000 entries in a text file
SP SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
G711 Sites

url flash:e164-pattern-map.cfg
description Inbound DP via Called
incoming called e164-pattern-map 400
codec g711ulaw
! This is an example of the contents

of E164 patterns text file
stored in flash:e164-patternmap.cfg
9192002010
5101001010
4081001010
URI Based Dialing Overview

INVITE sip:user@xyz.com
INVITE sip:user@xyz.com
CUBE
Enterprise
abc.com
SBC
Enterprise
xyz.com
Existing CUBE behavior:

In CUBE URI based routing (user@host), the user part must be present and must be an
E164 number
The outgoing SIP Request-URI and To header URI are always set to the session target
information of the outbound dial-peer
For Req-URIs with same user name e.g. hussain@cisco.com, hussain@google.com, two
different dial-peers are configured with the respective session targets
URI Based Dialing Enhancement

URI Pass Through
INVITE sip:1234@cisco.com
incoming uri request 1
voice-class sip call-route url
For Your
Reference
CUBE
INVITE sip:1234@cisco.com
destination uri 1
voice-class sip requri-passing
voice class uri 1 sip

host cisco.com
By default, the host portion is replaced with the session target value of the matched
outbound dial-peer
Enhancement : Outgoing INVITE has same request URI as received in Incoming INVITE.
This can be achieved by configuring requri-passing in the outgoing dial-peer or
globally.
Allows for peer-to-peer calling between enterprises using URIs

User portion non-E164 format
INVITE sip:hussain@cisco.com
For Your
Reference
CUBE
INVITE sip:hussain@10.1.1.1
destination uri 1
host cisco.com
By default, alphanumeric/non-E164 users were not allowed

Enhancement : User part in Incoming INVITE Req-URI can be of Non-E164 format. e.g.
sip:hussain@cisco.com. Outgoing INVITE will have user portion as it is received i.e.
hussain (unless SIP profiles are applied).
Useful for video calls

User portion absent
INVITE sip:cisco.com
For Your
Reference
CUBE

INVITE sip:cisco.com
destination uri 1
voice-class sip requri-passing

host cisco.com
By default, call is rejected with 400 Bad Request
Enhancement : Incoming INVITE with no user portion (e.g. sip:cisco.com.) is supported. Dial-peer
matching will happen based on host portion. Outgoing INVITE Req-URI will not have any user portion
in this case (unless sip-profiles are applied).
If user portion is present in incoming INVITE To header, it is retained in outgoing INVITE To Header
If voice-class sip requri-passing is not configured, INVITE will go out as sip:10.1.1.1
REFER and 302, both consume and pass-through cases supported as well

Deriving Target host from Incoming INVITE Req-URI
INVITE sip:hussain@cisco.com
CUBE
INVITE sip:hussain@10.1.1.1 Skype

destination uri 1
Facebook Video

session target sip-uri
user hussain
user .*
For different hosts with the same user, multiple outgoing dial-peers had to be configured
Enhancement : To support URIs with the same user portion but with different domains, only
one dial-peer per can be configured. Outgoing dial-peer needs to be configured with session
target sip-uri instead of regular session target configuration. This will trigger DNS
resolution of the domain of incoming INVITE Req-URI and dynamically determine the session
target IP.
CUBE Call Admission Features
Call Admission Control at the edge...

CUBE provides various CAC mechanisms to safeguard your network from SIP based attacks and to enforce policies based on:
Maximum connections per destination
Total calls
Dial-peer or interface bandwidth
CPU & Memory
Call spike detection
Total Calls,
CPU, Memory
High Water Mark

Low Water Mark
CUBE
Call Spike
Detection
call spike call-number [steps

number-of-steps size milliseconds]
call spike 10 steps 5 size 200
CUBE
call threshold global [total/mem/cpu] calls low xx high yy

call treatment on
Max Calls per

Destination
Call #1
Max Bandwidth
based
Call #3 Rejected
by CUBE
Call #1 80Kbps
Call #2 80 Kbps
Call #2
Call #3
Call #3
Rejected by
CUBE
If a call spike is detected,

reject calls
CUBE

max-conn 2
Call #3 80 Kbps

max-bandwidth 160
CUBE
Call Admission Control at the edge...

CUBE provides various CAC mechanisms to safeguard your network from SIP based attacks and to enforce policies based on:
Maximum connections per destination
Total calls
Dial-peer or interface bandwidth
CPU & Memory
Call spike detection
Total Calls,
CPU, Memory
High Water Mark

Low Water Mark
CUBE
Call Spike
Detection
call spike call-number [steps

number-of-steps size milliseconds]
call spike 10 steps 5 size 200
CUBE
call threshold global [total/mem/cpu] calls low xx high yy

call treatment on
Max Calls per

Destination
Call #1
Max Bandwidth
based
Call #3 Rejected
by CUBE
Call #1 80Kbps
Call #2 80 Kbps
Call #2
Call #3
Call #3
Rejected by
CUBE
If a call spike is detected,

reject calls
CUBE

max-conn 2
Call #3 80 Kbps

max-bandwidth 160
CUBE
CUBE Call Center Features
Mid-call codec renegotiation

G.711
3
CVP
G.711
Call Xfer (signaling only)
Provider supports both

G.711 and G.729 codecs
G.729 /
G.711
SP SIP
SIP
CUBE
4
G.729
G.729
Call arrives on G.729 SIP trunk
CVP connects call to speech recognition server that

requires G.711 so the call renegotiates G.711 e2e
CVP xfers call to a remote agent that uses G.729
Call renegotiates to G.729 e2e
Mid-call Xcoder Insert/Drop

G.711
3
CVP
Transcoder Inserted
G.711
Call Xfer (signaling only)
Provider supports only

G.729 codec
G.729 /
G.711
SP SIP
SIP
CUBE
4
G.729
G.729
Transcoder Dropped
Call arrives on G.729 SIP trunk
CVP connects call to speech recognition server that

requires G.711. Since provider does not support G711
CUBE inserts transcoder
CVP xfers call to a remote agent that uses G.729
CUBE drops xcoder and e2e call becomes G.729 again
REFER Handling for Contact Centers

Enables CUBE to handle REFER messages more efficiently in contact center deployments
CUBE can operate in either consume mode or pass-through mode
REFER Consumption
A
3. INVITE
SIP SP
CUBE
2. INVITE
CVP
Based on Refer-To header,

CUBE does outbound dial-peer
match and sends out an INVITE
message
No supplementary-service sip refer
supplementary-service media-renegotiate
1. REFER
REFER Pass-through (Default mode)

A
SIP SP
CUBE
2. REFER
CVP
1. REFER
CUBE will pass across the

Refer message as-is without
any modification
REFER Handling Enhancement

A new CLI, refer consume, has been added to the SIP dial peer.
The final decision to consume or pass-through REFER is determined based on this new
CLI option configured on the Refer-To dial-peer.
supplementary-service sip refer
refer consume
Configured globally or
at inbound dial-peer
Configured at dialpeer that matches

refer-To
Yes (default)
No (default)
REFER Pass-through
Yes (default)
Yes
REFER Consume
No
No (default)
REFER Consume
No
Yes
REFER Consume
Outcome
Use Case: Contact Center and IVR Improvements

Call progress analysis (CPA) on SIP trunks supports an allSIP environment (inbound and outbound).
SIP-based CPA enables Cisco Outbound Call Center
solutions to support SIP trunk connections through CUBE.
SIP Dialer
Sent:
Received:
INVITE sip:2776677@9.41.35.205:5060
SIP/2.0
UPDATE
sip:sipp@9.42.30.151:7988;transport=UDP
SIP/2.0
Via: SIP/2.0/UDP
SIP/2.0/UDP 9.41.35.205:5060;branch=z9hG4bK6F26CF
9.42.30.151:7988;branch=z9hG4bK-16368-1-0
Via:
..
.
event=detected
--uniqueBoundary
status=Asm
Content-Type: application/x-cisco-cpa
pickupT=2140
Content-Disposition: signal;handling=optional
maxActGlitchT=70
numActGlitch=12
Events=FT,Asm,AsmT,Sit
valSpeechT=410
CPAMinSilencePeriod=608
maxPSSGlitchT=40
CPAAnalysisPeriod=2500
numPSSGlitch=1
CPAMaxTimeAnalysis=3000
silenceP=290
CPAMaxTermToneAnalysis=15000
termToneDetT=0
CPAMinValidSpeechTime=112
noiseTH=1000
actTh=32000
SIP SP
CVP
Contact Center
CUBE
Dialer will then instruct

CUBE on whether to
connect the call to an agent
or disconnect the call by
sending REFER, RE-INVTE,
BYE, CANCEL etc.
CUBE detects fax or voicemail tone
Transcoder inserted
to detect tones
CUBE will then
connect/disconnect the
call appropriately
Configuration on CUBE:
voice service voip
cpa
dspfarm profile 1 transcode universal
call-progress analysis
CUBE Call Recording
Use Case: Automated CUCM Voice / Video Recording

Delivering the industrys most flexible call recording architecture
SP IP Network
SP IP Network
CUBE
Enterprise Network
CUBE
CUBE
CUBE
Distributed
Recording
Distributed
Recording
CUCM
Centralized
Recording
Enhanced control CUCM has policy control over media forking on CUBE & GWs.
Better bandwidth utilization use any CUCM, gain selectivity in call forking.
Flexibility distributed or centralized architecture, use any vendors media recording.
Improved compliance - record even network-connected mobile devices.
Use Case: Standards-Based Recording

CUBE now supports the FINAL
SIPREC Draft (v17)
Partner Application
Cisco MediaSense
(authentication disabled w/o UCM)
MediaSense
CUBE supports early SIPREC

drafts with MediaSense.
SIP
RTP
SIP
SIP
SP SIP
RTP
CUBE
Call control (IP-PBX) independent

Configured on a per dial-peer level to fork RTP
RTP
CUBE-ENT as Part of HCS Architecture

November-2015
Physical deployment of Cube-Ent(With Multi VRF)

PSTN
Provider
Tenant-A
Tenant-B
MPLS
CUBE
ASA-FW-A
ASA-FW-B
N7k
PSTN CPE
Breakout
10GB link
CUBE
ASR
1000
Cube-Ent-A
(Act/Stby)
CUBE
1GB link
PSTN
Provider
FI-62xx
1GB link
N100V
UCS
Chassis
UCAPPS-Tenant-A
UCAPPS-Tenant-B
ASR
1000
Cube-Ent-B
(Act/Stby)
Logical View of a HCS Customer/Tenant

HCS Management
Apps
UC-App Customer-A
(Vlan inside)
(Default GW = HSRP-cust-a-inside)
vlan-cust-A(inside)
Mgmnt Vlan
ASA-FW
NAT
vlan-custA(inside)
FW-Context
CustA
vlan-custA(outside)
Aggregation Layer Switch A

Cust-A-inside-VRF
vlan-cust-A(inside)
vlan-Cust-A(inside)
HSRP-Cust-a-inside
Static route to ASA for all traffic out of
vlan
Cust-A-Outside-VRF
vlan-cust-A(outside)
vlan-Cust-A(outside)
Hsrp-Cust-a-outside
Static route to AS for UC Apps
Static route to CUBE-Ent for PSTN
Static route to ASA for Mgmnt
To Agg-B(Peer)
PE
PE
CE
Customer
Premise
MPLS Core
CUBE-Ent
Cust-A-VRF
vlan-custA(outside)
SBC
(sbc inside
adjacency/i
nterface)
2 HSRPs
2 Vlans
4 Static routes
2 VRF
3 BGP peers
HCS SBC/CUBE-Ent Redundancy Options

In Chassis Redundancy
Network
Network
ASR
1000
Redundant control plane and

media/forwarding plane
Active/standby RP and ESP
<1ms failover
No traffic impact on most software
upgrades
customers
Network
Chassis to Chassis Redundancy

Recommended option
Keep alives
ASR
1000
ASR
1000
customers
Dual Chassis
Network
One chassis is active and

other is standby
Standby chassis takes over
sessions when active fails
No traffic impact on software
upgrades
DNS or call routing shares

call load over chassis
Traffic can be moved
between chassis for
software upgades
Not deployed for HCS

ASR
1000
ASR
1000
customers
No Redundancy
Network
ASR
1000
simplex control plane and

media/forwarding plane
Single RP and ESP
Traffic is impacted on software
upgrades
customers
HCS SBC/CUBE-Ent Inter-Chassis Redundancy

LAN Connectivity
Drawing based on DC Connection Option 1

-
Data and Control traffic on different subint of the same PC po300
Control link used to communicate the status of the routers, data link used to transfer stateful
information from the SBC.
PC po300 using different source slots
One interface used for PSTN facing SIP and RTP traffic
One interface used for Customer facing SIP and RTP traffic
HCS SBC/CUBE-Ent Routing Logic

Dial
peer
Dial
peer
Dial
peer
Dial
peer
Drawing based on CUBE-Ent interfacing with CS2K for PSTN

-
Per customer: as many adjacency as call processing CUCM and as many

adjacency as PSTN SIP interfaces
Routing logic is purely based on adjacency (no number manipulation)
Per customer: On each set of adjacency, incoming routing logic is

sending call to the other set of adjacency (i.e. from CUCM to PSTN and
vice versa)

CUBE-EnT Preso For HCS Deployathon - V5

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CUBE(ENT) as Replacement for

CUBE(SP) & MetaSwitch Perimeta

Market Overview & CUBE Market Positioning

CUBE-ENT relevance to Service Providers

Introducing mVRF and mTenancy on CUBE-ENT

CUBE-ENT features relevant to HCS

CUBE-ENT features compared to CUBE-SP

HCS Solution Architecture with CUBE-ENT

CUBE-ENT as Part of Cisco Call Center Solution (OPTIONAL)

CUBE-ENT voice security policy solutions (OPTIONAL)

CUBE-SP to CUBE-ENT license migration.

SIP Trunk Market Trends

NORTH AMERICA: Per FCC, TDM to SIP trunk

SIP Session Deployment Growth

Up from 30% at end of FY14.

Accelerating going forward into CY16 thru CY19.

EUROPE: Huge transition is now underway in

AJPC: Asia, except Australia and Japan, are at

Total Market: Worldwide, less than 10% of SIP trunk

SPs are (desperately) looking for ways to

Cumulative Total: ~65 M sessions

Remaining Market Transition: ~630 M sessions

Telco Customer Strategies

Cisco Unified Border Element (CUBE)

Rich Media (Voice, Video Content) Rich Media

Proven Interoperability and

Validated with Service

Cisco Interoperability Portal:

CUBE & Cisco UCM Session Management Edition (SME)

Centralizes dial plan

CUBE: Primary Strategic Differentiators

Solutions for the

Enables flexible deployment models:

Virtualized CUBE (with CUSP load balancing)

64,000 sessions enabled by

800 ISR Capacity: 100600

Active Voice Call (Session) Capacity

CUBE-Enterprise Value Add Use Cases - Examples

IVR & CPA

Moderate- High Cost

Enterprise SBC Market

CUBE: Enterprise SBC Market Leadership

CUBE Enterprise Intellectual Property is 100% owned by Cisco.

CUBE Primary Differentiators:

Router Integration (see chart)

Price / Performance Scalability

CUBE adds value to Cisco UC Solutions:

Voice Security Policy

Major CUBE Customers:

Royal Bank of Scotland

<5 800 ISR

Virtualized CUBE (with CUSP load

CUBE Enterprise Training Labs

CUBE Enterprise On-Line Training:

CUBE Technical Resources:

SIP Services Operational Vectors faced by

CUBE Strategic Value to SPs for SIP Services:

Efficient Network Architecture for SIP Services

Efficient Network Architecture for SIP Services

Customer Prem SBC

Reduces risk of Aggregation SBC Failure from unusual SIP normalization

Efficient Network Architecture for SIP Services

Customer Prem SBC