Discussion Agenda
• CUBE & CUSP Market Position and Product Strategy
• CUBE & CUSP Enhancements from Recent Releases (past 18 months)
• CUBE & CUSP Enhancements in Most Recent Release
• CUBE & CUSP Futures (A Sneak Peek)
• Deployed by over 17,000 organizations, each with at least 200 session license in use.
• Over 10 milllion licensed SIP sessions
• Approximately 1000 new customers per quarter.
• Deployed in 160+ countries
• Diverse Channels with broad range of VARs & SP partners
• In market and continuously enhanced for over 12 years.
• Identified as market leading Enterprise SBC by Infonetics for the past 3 years (27% share in 2014)
CUBE Adoption Market Statistics
Estimated Market Growth in SIP Sessions
Worldwide per Infonetics 2014
Cisco Interoperability Portal:
www.cisco.com/go/interoperability
Validated with service providers world-wide
Tested with 3rd party PBXs
Standards based
CUBE Interoperability Proven Interoperability and Interworking with Service Providers Worldwide
SESSION
CONTROL
Call Admissions
Control
Trunk Routing
Ensuring QoS
Statistics and Billing
Redundancy/
Scalability
INTERWORKING
SIP - SIP
H.323 - SIP
SIP Normalization
DTMF/ PT
Interworking
Transcoding
Codec Filtering
DEMARCATION
Fault Isolation
Topology Hiding
Network Borders
L5/L7 Protocol
Demarcation
SECURITY
Encryption
Authentication
Registration
SIP Protection
Voice Policy
Firewall Placement
Toll Fraud
Enterprise 1
IP SIP
CUBE
IP Enterprise 2
IP SIP
Rich Media (Real time Voice & Video) Rich Media
Cisco Unified Border Element (CUBE) Router-based or virtual Session Border Control (SBC)
6
CUBE CUBE
SBC Integration on the Router
• Leverages installed base and knowledge base
Broadest Scale of price
performance
• Solutions for the smallest to largest deployments
Integrated SBC and TDM Gateway
• Simplifies transition from to IP PSTN
Voice
Security Policy
• TDOS protection with granular policy-based enforcement
Integration with Cisco
Collaboration
• Cisco Unified CM recording solutions
• CVP call center solutions
• WEBEX integration for Cloud Connect Audio
CUBE: Primary Strategic Differentiators Unmatched by competitors’ SBCs
Enables flexible deployment models:
centralized, distributed, hybrid
E A B C D
CUBE: Platform Portfolio Unequaled Price / Performance SBC Scalability
Call
s p
er
Se
co
nd
<5
8–12
50–150
17
20–35
ASR 1004/6 RP2
2900 ISR 100-600
1001-X & 1002-X
ASR
800 ISR Up to 50
Active Voice Call (Session) Capacity
64,000 <50 500–600 1,000 2,500 6,000 12,000 4 16,000
ASR 1006 Highest Transcoding Capacity:
9,000 G729 to G711 Calls
ASR 1001 Highest Density
10,000 Session in 1 RU 4300 ISR 100-1000
4400 ISR 3000-6000
3900 ISR 800-2,500
4400 ISR
Introduced in
July 2013
ASR 1002-X
Introduced in
July 2012
ASR 1001-X
Introduced in
March 2014
4300 ISR
Introduced
in Sept 2014
NanoCUBE
Oct 2013
CUBE Market Leading Scalability Only SBC Platform to Extend Across All Customer Market Segments
Mid-Market
Commercial 100 to 3000 sessions
EdgeWater
Adtran
ACME / Oracle
Sonus
CISCO CUBE ISR-G2
ISR 43XX / 44XX
SMB <100 sessions
Enterprise >3000 Sessions
CISCO CUBE ISR 88X
SPIAD 29XX
CISCO CUBE ASR 1006
ISR 43XX / 44XX
ISR-G2
Audio Codes
InGate
Avaya
Customer
Segment
Platforms SIP service Direct VAR SP
Large
Enterprise
Mostly ASR
Some ISR
Trunk
Lineside (remote
worker)
XXX XX
DoD ASR and ISR Trunk and
Lineside
XXX
Small
Enterprise
Mostly ISR
Some ASR
Trunk
Lineside (HCS)
XXX XXX XXX
Large
Commercial
ISR Trunk
Lineside
(HCS)
XXX XXX
SMB Cisco 8XX Trunk
Lineside
(Broadsoft)
XX
CUBE Market Segmentation Opportunity Ranking by Channel
CUBE Architecture Flexibility Efficiently Supports All SIP Architectures for Voice or Video Services
Enterprise IP WAN
Distributed SIP Architecture Centralized SIP Architecture
Hybrid SIP Architecture
Enterprise IP WAN
IP PSTN
Enterprise IP WAN
CUBE
CUBE CUBE CUBE CUBE
IP PSTN
IP PSTN
CUBE CUBE
CUBE
CUBE CUBE CUBE
How to Choose a SIP Trunk Architecture Collaboration services should determine the architecture type
Collaboration Service Centralized Distributed Hybrid
Audio only: 1 to 1 Best Good Better
Audio only: multi-party conferencing Good Better Better
Audio & Video: 1 to 1 Good Better Better
Audio & Video: multi-party conferencing Worst Best Better
Cloud Collaboration Worst Best Better
CUBE enables WEBEX Cloud Connect Audio (CCA) Practical Application of Distributed SIP Trunking
Requirements
• Replaces TDM audio connection to WEBEX with VOIP using SIP signaling.
• WEBEX cloud becomes a portal off of Enterprise WAN
How
• CUBE Reduces SIP protocol “chatter” between IP-PBX and WEBEX cloud thru “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.
Benefit
• Dramatic savings thru elimination of TDM, plus excellent conference experience thru efficient network usage.
Enterprise
IP WAN
(MPLS)
Branch Office Branch
Office
Headquarters
A
CUBE
CUBE
CUBE
Branch Office
WEBEX
CUBE
TDM PSTN
CUBE
CUSP: Optimization of SIP Signaling Simplify, Normalize and Balance Call Signaling between all SIP Network Elements
IP-PSTN
VXML
IP-PSTN
VXML
Without CUSP SIP Proxy With CUSP SIP Proxy
CVP CVP
CUCM CUCM
CUBE CUBE CUBE CUBE
UCCE/X UCCE/X
Cisco Unified SIP Proxy CUSP
• PRIMARY FUNCTIONS:
• Stateless Call Routing
• Load Balancing
• SIP normalization
• PRIMARY BENEFITS:
• Reduced call flow complexity
• Increased call rate processing
• Signaling interoperability
• Enable increased system capacity
CUSP Strategic Use Cases 1. SBC Load Balancing -
• CUSP is Primary Recommendation
• SIP Proxy Function must reside in DMZ
2. Call Center Routing
• CUSP is Primary Recommendation
• “Stateless” SIP Proxy routing mechanism simplifies call center
3. Internal Call Routing
• CUSP is Secondary Recommendation
• Stateful call routing is performed by CUCM-Session Management Edition
• CUSP can be used as an SME-lite…..but is not a session manager
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
CUSP-SRE
CPS – RR On
CPS – RR Off
200
400
4:1
8:1
6:1
12:1
13:1
26:1
CUSP UCS-E
CPS – RR On
CPS – RR Off
750
1500
15:1
30:1
23:1
46:1
50:1
100:1
ISR G2 CUBE Ent ASR Parity
with ISR
ASR / ISR-4K*/vCUBE (CSR)*
CUBE Vers.
2900/ 3900 FCS CUBE Vers.
IOS XE Release FCS
9.0.1 15.3.1T Oct 2012 >95% 9.0.1 3.8 15.3(1)S Oct 2012
9.0.2 15.3(2)T Mar 2013 >95% 9.0.2 3.9 15.3(2)S Mar 2013
9.5.1 15.3(3)M1 Oct 2013 >95% 9.5.1 3.10.1 15.3(3)S1 Oct 2013
10.0.0 15.4(1)T Nov 2013 >95% 10.0.0 3.11 15.4(1)S Nov 2013
10.0.1 15.4(2)T Mar 2014 >95% 10.0.1 3.12 15.4(2)S Mar 2014
10.0.2 15.4(3)M July 2014 >95% 10.0.2 3.13 15.4(3)S July 2014
10.5.0 15.5(1)T Nov 2014 >95% 10.5.0 3.14 15.5(1)S Nov 2014
11.0.0 15.5(2)T Mar 2015 >95% 11.0.0 3.15 15.5(2)S Mar 2015
11.1.0 15.5(3)M July 2015 >95% 11.1.0 3.16 15.5(3)S July 2015
11.5.0 15.6(1)T Nov 2015 >95% 11.5.0 3.17 15.6(1)S Nov 2015
12.0.0 15.6(2)T Mar 2016 >95% 12.0.0 3.18 15.6(2)S Mar 2016
12.1.0 15.6(3)M July 2016 >95% 12.1.0 3.19 15.6(3)S July 2016
20 * IOS-XE3.13.1 or later recommended for all ISR-4K series and XE3.15 for vCUBE
CUBE Software Release Mapping
Feature Benefit Strategic
Differentia-
tor
Enhanced Call Routing Simplified and enhanced dial peers for advanced call
routing
URI-based dialing Support for optional dialing mechanisms, such as
with MSFT LYNC
SIP-based proxy registration with
3rd party call control
Support standardized remote registration for hosted
call control services
Integration with Cisco Unified CM
10.0 Recording Solution More deployment options
Selective Event Tracing features
for traffic debug and analytics
Improved serviceability to simplify analysis of
signaling anomalies.
Support for Cisco ISR 4000 Series Improved price performance scalability
CUBE Enhancements from Recent Releases
E
C
Enhanced Call Routing - Destination Server Group
• Enables multiple destinations (session targets) to be grouped and applied to a single outbound dial-peer
• The outbound dial-peer routing an outgoing call can use the Destination Server Group to select the session target based on either round robin or preference logic
• Avoids configuration of multiple dial-peers with the same capabilities but different destinations. E.g. Multiple subscribers in a cluster
2
2
voice class server-group 1
hunt-scheme {preference | round-robin}
ipv4 1.1.1.1 preference 5
ipv4 2.2.2.2
ipv4 3.3.3.3 port 3333 preference 3
ipv6 2010:AB8:0:2::1 port 2323 preference 3
ipv6 2010:AB8:0:2::2 port 2222
* DNS target not supported in server group
dial-peer voice 100 voip
description Outbound DP
destination-pattern 1234
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
session server-group 1
Multiple Destination-Patterns Under Same Outbound Dial-Peer
2
3
SIP Trunk SP SIP Trunk
CUBE
IP PSTN A
(408)100-1010
(510)100-1010
(919)200-2010
(919)200-2000
(510)100-1000
(408)100-1000
voice class e164-pattern-map 100
e164 919200200.
e164 510100100.
e164 408100100.
dial-peer voice 1 voip
destination e164-pattern-map 100
codec g729r8
session target ipv4:10.1.1.1
voice class e164-pattern-map 100
url flash:e164-pattern-map.cfg
dial-peer voice 1 voip
destination e164-pattern-map 100
codec g711ulaw
session target ipv4:10.1.1.1
! This is an example of the contents
of E164 patterns text file stored in flash:e164-pattern-map.cfg
9192002010 5101001010 4081001010
Site A
Site B
Site C
Site A
Site B
Site C
G729 Sites
G711 Sites
Provides the ability to combine multiple
destination-patterns targeted to the
same destination to be grouped into a
single dial-peer
Multiple Incoming Patterns Under Same Incoming Dial-peer
2
4
SIP Trunk SP SIP Trunk
CUBE
IP PSTN A
(408)100-1010
(510)100-1010
(919)200-2010
(919)200-2000
(510)100-1000
(408)100-1000
voice class e164-pattern-map 100
e164 919200200.
e164 510100100.
e164 408100100.
dial-peer voice 1 voip
description Inbound DP via Calling
incoming calling e164-pattern-map 100
codec g729r8
voice class e164-pattern-map 200
url flash:e164-pattern-map.cfg
dial-peer voice 2 voip
description Inbound DP via Called
incoming called e164-pattern-map 200
codec g711ulaw
! This is an example of the contents
of E164 patterns text file stored in flash:e164-pattern-map.cfg
9192002010 5101001010 4081001010
Site A
Site B
Site C
Site A
Site B
Site C
G729 Sites
G711 Sites
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 dial-
peers required with the same routing
capability
Destination Dial-peer Group
FEATURE DESCRIPTION
• Enables grouping of outbound dial-peers based on an incoming dial peer match.
• Specific outbound dial-peers are selected and associated with a new dial peer group CLI:
“voice class dpg <tag>”.
• The inbound VOIP dial-peer references the dial peer group with a new CLI:
“destination dpg <tag>”
SUMMARY OF BENEFITS
• Simplifies dial plan routing logic for associating inbound to outbound dial peers. • Eliminates need to configure extra outbound dial-peers that are sometimes needed to achieve desired call routing
outcome
• Potentially reduces the overall number of dial-peers evaluated per call, thereby increasing processor efficiency.
2
5
Destination Dial-peer Group Configuration
voice class dpg 10000
description Voice Class DPG for DP Source SJ
dial-peer 1001 preference 2
dial-peer 1002 preference 1
dial-peer 1004 preference 3
!
!
dial-peer voice 100 voip
description DP Source SJ w/voice class dpg
incoming called-number 13..
destination dpg 10000
dial-peer voice 1001 voip
description DPG 10000
destination-pattern 1341
session protocol sipv2
session target ipv4:10.1.1.1
!
dial-peer voice 1002 voip
description DPG 10000
destination-pattern 13..
session protocol sipv2
session target ipv4:10.1.1.2
!
dial-peer voice 1003 voip
description DPG 10000
destination-pattern 134.
session protocol sipv2
session target ipv4:10.1.1.3
!
dial-peer voice 1004 voip
description DPG 10000
destination-pattern 1...
session protocol sipv2
session target ipv4:10.1.1.4
!
1. Incoming Dial-peer
is first matched
2. Now the DPG associated
with the INBOUND DP is
selected
OUTBOUND
INBOUND
Incoming Dialed #
1341
URI Based Dialing Overview
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. [email protected], [email protected], two different dial-peers are configured with the respective session targets
Enterprise
xyz.com Enterprise
abc.com
CUBE SBC
INVITE sip:[email protected]
INVITE sip:[email protected]
URI Based Dialing Enhancement – URI Pass Through
• 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
3
0
INVITE sip:[email protected]
INVITE sip:[email protected] CUBE
dial-peer voice 100 voip
incoming uri request 1 dial-peer voice 200 voip
session protocol sipv2 destination uri 1
voice-class sip call-route url session protocol sipv2
session target ipv4:10.1.1.1
voice-class sip requri-passing
voice class uri 1 sip
host cisco.com
For Your
Reference
URI Based Dialing Enhancement – ‘User’ portion non-E164 format
• 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:[email protected]. Outgoing INVITE will have user portion as it is received i.e. ‘hussain’ (unless SIP profiles are applied).
• Useful for video calls
3
1
INVITE sip:[email protected]
INVITE sip:[email protected] CUBE
dial-peer voice 100 voip
incoming uri request 1 dial-peer voice 200 voip
session protocol sipv2 destination uri 1
voice-class sip call-route url session protocol sipv2
session target ipv4:10.1.1.1
voice class uri 1 sip
host cisco.com
For Your
Reference
URI Based Dialing Enhancement – ‘User’ portion absent
• 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
3
2
INVITE sip:cisco.com
INVITE sip:cisco.com CUBE
dial-peer voice 100 voip
incoming uri request 1 dial-peer voice 200 voip
session protocol sipv2 destination uri 1
voice-class sip call-route url session protocol sipv2
session target ipv4:10.1.1.1
voice-class sip requri-passing
voice class uri 1 sip
host cisco.com
For Your
Reference
URI Based Dialing Enhancement – Deriving Target host from Incoming INVITE Req-URI
• 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.
3
3
INVITE sip:[email protected]
INVITE sip:[email protected] CUBE
dial-peer voice 100 voip
incoming uri request 1 dial-peer voice 200 voip
session protocol sipv2 destination uri 1
voice-class sip call-route url session protocol sipv2
session target sip-uri
voice class uri 1 sip
user hussain
user .*
Skype
Facebook Video
NanoCUBE Deployment Scenarios
IAD
IP PBX TDM PBX CUCM
CUBE
NANO
-CUBE 8xx
NANO-CUBE
8xx
SIP
PRI SIP
SIP CPE SIP
Hosted Service
Small Business
SIP
SIP Trunking
Small Business
PRI To SIP
Enterprise
NanoCUBE
Hosted Service Small
Business
SIP Trunking Small
Business
Service Provider Call Control
3
4
CUBE Value Proposition to SP’s: Breadth of Scalability increases SP market size and reduces operation costs
SMB Large
Enterprise
Hosted Services
SIP Line-side Call control in the cloud
Managed Services
SIP TRUNK Call control on Customer Prem
Mid-range SBC
High
Capacity
SBC
Low end
SBC’s
CUBE
Enterprise Network CUBE
CUCM
• 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, uses any vendors media
recording
• Improved Compliance - record even network-connected mobile devices
CUBE
CUBE
CUBE
SP IP Network
Dynamic CUCM Triggered Call Recording Delivering Industry’s Most Flexible Call Recording Architecture
Centralized
Recording
Distributed
Recording
Distributed
Recording
SP IP Network
Debugging Made Easier
3
7
Router# debug ccsip level <critical | info | notify | verbose>
Categorize Debugs based on Severity
Existing SIP debugs have become too verbose and un-manageable. To minimize verbosity, the SIP-INFO debugs are further categorized based on functionality and Level
Categories only applicable when CCSIP INFO or ALL debug is enabled
Categorization based on Severity
1. Critical
2. Notifications
3. Informational
4. Verbose
Severity
Level Description
1 Critical Feature specific Errors, things going wrong,
resource failures that does not fail call as such
2 Notifications Important milestones reached. Important steps
while processing that needs to be noticed
3 Informational Much of the details to understand flow. These
give more information related to working of flow
4 Verbose Information that is in too detail and not really
much helpful in debugging
Debugging Made Easier
3
8
CUBE# show cube debug category codes
Categorize Debugs based on Functionality
This CLI is used to collect the predefined debug features category codes , which helps in analysis of debugs manually.
|-----------------------------------------------
| show cube debug category codes values.
|-----------------------------------------------
| Indx | Debug Name | Value
|-----------------------------------------------
| 01 | SDP Debugs | 1
| 02 | Audio Debugs | 2
| 03 | Video Debugs | 4
| 04 | Fax Debugs | 8
| 05 | SRTP Debugs | 16
| 06 | DTMF Debugs | 32
| 07 | SIP Profiles Debugs | 64
| 08 | SDP Passthrough Deb | 128
| 09 | Transcoder Debugs | 256
| 10 | SIP Transport Debugs | 512
| 11 | Parse Debugs | 1024
| 12 | Config Debugs | 2048
| 13 | Control Debugs | 4096
| 14 | Mischellaneous Debugs| 8192
| 15 | Supp Service Debugs | 16384
| 16 | Misc Features Debugs| 32768
| 17 | SIP Line-side Debugs | 65536
| 18 | CAC Debugs | 131072
| 19 | Registration Debugs | 262144
|-----------------------------------------------
Debugging Made Easier
3
9
Router# debug ccsip feature < audio | cac |
config | control | dtmf | fax | line | misc |
misc-features | parse | registration | sdp-
negotiation | sdp-passthrough | sip-profiles
| sip-transport | srtp | supplementary-
services | transcoder | video >
Categorize Debugs based on Functionality
Categorization based on Functionality
1. Audio/video/sdp/control
2. Configuration /sip-transport
3. CAC
4. DTMF/FAX/Line-side
5. Registration
6. Sdp - passthrough
7. Sip-profile/SRTP/transcoder
Example: enabling DTMF and audio debugs only with default log level is considered.
CUBE#sh debugging
CCSIP SPI: SIP info debug tracing is enabled (filter is OFF)
CCSIP SPI: audio debugging for ccsip info is enabled (active)
CCSIP SPI: dtmf debugging for ccsip info is enabled (active)
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/verbose/32/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing
negotiated dtmf = 0,
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/info/2/sipSPIUpdateCallEntry:
Call 444 set InfoType to SPEECH
DTMF(32) debug code
Audio(2) debug code
Feature Benefit Strategic
Differentia-
tor
Virtualization of both CUBE and
CUSP
Enable even greater flexibility in CUBE deployment
models, including CUBE Clustering
SIP-based Call Progress Analysis Allow Outbound Call Center over a SIP PSTN E
Pass thru of all headers on SIP
mid-session message types
Allow end point to end point processing of mid-call
signaling
HA checkpoint feature for
transcoded audio and DTMF
Improved High Availability for Advanced Media
Features
Multi-M Line Enhancements &
Video Forking
Enable advance multi-media support and video
recording with WEBEX & Remote Expert E
Provisioning & Monitoring of
CUBE with Prime Collab Integrate CUBE management with Prime Collab D
Voice Security Policy
Performance Enhancements
Enable use of voice policy even at the very highest
scale of call connections ( >10,000 sessions)
E
CUBE Enhancements Being Released in April
E
D
E
C
E
Cisco Unified SIP Proxy (CUSP) A Critical Element of Cisco Collaboration Infrastructure
IP-PSTN
VXML
IP-PSTN
VXML
Without CUSP With CUSP
CVP CVP
CUCM CUCM
CUBE CUBE CUBE CUBE
UCCE/X UCCE/X
Optimize, normalize, balance call signaling between all SIP elements
• Virtualized CUSP 9.0 (available today) • Virtualized support for CUSP for long term platform
• Transition to Smart Licensing & SWSS
• Enhance CUSP SNMP monitoring & serviceability (version 9.1)
• Virtualized CUBE option (April) • IOS XE 3.15
• All CUBE features except those supported by DSP
• CUBE clustering with dynamic activation & deactivation of virtualized CUBE
Announcing CUBE & CUSP Virtualization Enhanced Deployment Flexibility and Scalability through Virtualization
UCS Servers
Introducing vCUBE (CUBE on CSR 1000v) Virtual Architecture
• CSR (Cloud Services Router) 1000v runs on a Hypervisor – IOS XE without the router
CSR 1000 (virtual IOS XE)
Console Mgmt ENET Ethernet NICs Flash / Disk Memory Virtual CPU
RP (control plane)
Chassis Mgr.
Forwarding Mgr. IOS XE
Kernel (incl. utilities)
ESP (data plane) Chassis Mgr.
Forwarding Mgr.
QFP Client / Driver
FFP code
Hypervisor
Hardware
vSwitch NIC
GE GE … X86 Multi-Core
CPU Memory Banks
ESXi Container
CUBE signaling CUBE media processing
Introducing vCUBE (CUBE on CSR 1000v) • CSR1000v is a virtual machine, running on x86 server (no specialized hardware) with physical
resources are managed by hypervisor and shared among VMs
• Can be installed either using an OVA file or deployed with an ISO image
• Requires APPX or AX CSR licensing package to access voice CLI and increase throughput from 100 kbps default.
• CUBE Licensing follows ASR1K SKUs and currently is still trust based
• No DSP based features (transcoding/inband-RFC2833 DTMF/ASP/NR) available
• vMotion for vCUBE not supported today
• vCUBE Tested Reference Configurations [UCS base-M2-C460, C220-M3S, ESXi 5.1.0 & 5.5.0]
ASR, CSR & ISR-G2/4K Feature Comparison
4
5
General SBC 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/Co-
existence 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 Not Available
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 and CUBE co-
existence Not Available Exists XE3.11 Roadmap
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: Platform Portfolio Unequaled Price / Performance SBC Scalability
Call
s p
er
Se
co
nd
<5
8–12
50–150
17
20–35
ASR 1004/6 RP2
2900 ISR 100-600
1001-X & 1002-X
ASR
800 ISR Up to 50
Active Voice Call (Session) Capacity
64,000 <50 500–600 1,000 2,500 6,000 12,000 4 16,000
ASR 1006 Highest Transcoding Capacity:
9,000 G729 to G711 Calls
ASR 1001 Highest Density
10,000 Session in 1 RU 4300 ISR 100-1000
4400 ISR 3000-6000
3900 ISR 800-2,500
ASR +
virtualized CUSP 64,000 sessions enabled by CUBE
clustering
1006 ASR & CUSP
vCUBE to be added
April 2015
4400 ISR
Introduced in
July 2013
ASR 1002-X
Introduced in
July 2012
ASR 1001-X
Introduced in
March 2014
4300 ISR
Introduced
in Sept 2014
New Support for very large deployments with virtualized CUSP
• Integrates with CUBE to support CUBE clusters
• Delivers 4x more call capacity than CUBE alone
• Enables CUBE Clusters to:
• Increase scalability of all CUBE features
• Provide reliability via data center redundancy
• Support very large centralized SIP trunk deployments
CUBE: For Deployments of Every Size
Se
ss
ion
s
1
6
00
0
16
,00
0
64
,00
0
Very Large
Up to 64,000 calls
Large
Up to 16,000
calls
Small to
Medium
Up to 6000 calls
CUBE on ISR
or virtual CUBE on ASR
or virtual
CUBE on ASR or
virtual with CUSP 9.0
Call Progress Analysis (CPA) on SIP Trunks Cisco Call Center supports an ALL SIP Environment (inbound and outbound!)
4
8
CVP
SIP Dialer
SIP SP
CUBE CUBE detects FAX / Voice Mail tone
Configuration on CUBE:
voice service voip
cpa
dspfarm profile 1 transcode universal
call-progress analysis
Transcoder Inserted
to detect tones 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 will then
connect/disconnect the
call appropriately
Contact Center
SIP-based CPA enables Cisco Outbound
Call Center solution to support SIP trunk
Connections through CUBE
Sent:
UPDATE sip:[email protected]:7988;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 9.41.35.205:5060;branch=z9hG4bK6F26CF
…………….
event=detected
status=Asm
pickupT=2140
maxActGlitchT=70
numActGlitch=12
valSpeechT=410
maxPSSGlitchT=40
numPSSGlitch=1
silenceP=290
termToneDetT=0
noiseTH=1000
actTh=32000
Received: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 9.42.30.151:7988;branch=z9hG4bK-16368-1-0 ……………..
--uniqueBoundary
Content-Type: application/x-cisco-cpa
Content-Disposition: signal;handling=optional
Events=FT,Asm,AsmT,Sit
CPAMinSilencePeriod=608
CPAAnalysisPeriod=2500
CPAMaxTimeAnalysis=3000
CPAMaxTermToneAnalysis=15000
CPAMinValidSpeechTime=112
Prime Collaboration CUBE Provisioning
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname ${hostname}
!
logging message-counter syslog
logging buffered 51200 warnings
no logging console
!
voice service voip
allow-connections sip to sip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
rel1xx disable
header-passing error-passthru
early-offer forced
midcall-signaling passthru
sip-profiles 100
!
voice class codec 1
codec preference 1 ${codec-pref-1}
codec preference 2 ${codec-pref-2}
codec preference 3 ${codec-pref-3}
!
CUBE Monitoring
5
0
Area Information Method
Router Health CPU, Memory, I/f CISCO-PROCESS-MIB, cpmCPUTotal5minRev
CISCO-MEMORY-POOL-MIB, ciscoMemoryPoolTable
IF-MIB, IfEntry
SIP Trunk Status SIP Trunk Status SIP OOD Options Ping, CLI dial-peer status
Traffic Reports (Calls,
Sessions, Capacity Planning, Errors)
Trunk Utilization
CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallVolume
Older CUBE: DIAL-CONTROL-MIB, callActive
CISCO-DIAL-CONTROL-MIB, cCallHistoryTable
CUBE 8.5: SIP RAI Trunk Utilization
Call Arrival Rate CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallRateMonitor
Call Success/Failure DIAL-CONTROL-MIB, dialCtlPeerStatsSuccessCalls, dialCtlPeerStatsAcceptCalls,
dialCtlPeerStatsFailCalls, dialCtlPeerStatsRefuseCalls
CISCO-SIP-UA-MIB, cSipStatsErrClient, cSipStatsErrServer, cSipStatsGlobalFail
SIP retries CISCO-SIP-UA-MIB, cSipStatsRetry
Media Resources (DSPs)
DSP Availability CISCO-DSP-MGMT-MIB, cdspCardResourceUtilization, cdspDspfarmUtilObjects
Transcoding util. CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailTranscodeSess, cdspTotUnusedTranscodeSess
MTP utilization CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailMtpSess, cdspTotUnusedMtpSess
Voice Quality Loss, delay, jitter CISCO-VOICE-DIAL-CONTROL-MIB, cvVoIPCallActiveTable
IP SLA CISCO-RTTMON-RTP-MIB, rttMonJitterStatsTable , rttMonLatestJitterOperTable
More info in CUBE Management and Manageability Specification at: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550.html
For Your
Reference
Prime Collaboration - Assurance CUBE Features Benefits matrix
Features Benefits Monitoring Cisco Unified
Border Element
(CUBE)
Has built in knowledge to auto-discover the CUBE system.
Enable administrator to monitor CPU and DSP intensive tasks like
Transcoding and MTP session usage based on threshold alerts.
Detecting SIP trunk Outage Accurate Option Ping Method based CUBE SIP Trunk outage detection
Pro-actively Monitoring
SIP trunk Utilization Incoming or Outgoing Call stats to understand call traffic pattern
Incoming or Outgoing Utilization to understand trunk usage pattern
Detecting DSP failure Detects and notifies when a DSP chip/card fails that might potentially
cause service disruption such as call drop due to unavailability for
resources for transcoding.
Call Performance metrics Additional CUBE KPIs such as call stats for deeper monitoring
2014 TDoS
Warnings: • FBI
• DHS
• NENA 911
• APCO
Telephony Denial of Service (TDoS) Dramatic Increase in number and severity of TDOS attacks
Network Security (Malicious Packets)
Rogue RTP
Mal-Formed SIP Events
Untrusted Sources
Eavesdropping
Application Security (Malicious Calls)
Telephony DoS
Social Engineering / ATO
IRSF / Toll Fraud
Robocalls / Scams / Vishing
Harassing Calls
UC federation
INSPECT - SCORE (All calls at no cost)
TEST /
PROBE ENFORCE
• Real-Time Inspection of ALL calls
• Real-Time Baseline
• Meta-Data (Volume, Rate)
• E164 Derived Meta-Data (Google,
lnplookup)
• SIP Meta-Data (Headers, Call State)
• Customer Dips (White List)
• Real-Time Inspection of SOME calls
• Active Probing (Twilio, TrustID)
• 3/4 of TrustID value is in Active Call DIP
• Elastic
• Real-Time N-Factor of FEW calls
• Twilio “Turing Test”
• Elastic
• Fast MVP
100%
CUBE: Per Call Security Policy Process Each and every call is filtered and evaluated for threat potential in real time
~20% <5%
OTHER DB’s
Mobile Authentication
Landline Authentication
Substantial Managed Blacklisting
Location Authentication
SIP Quality and IP Anomalies
Assigned Numbers Valid Numbers Etc.
Voice Policy
Carrier TDM
Carrier SIP
Report, Record Reroute, Reject
CUBE
CUBE Voice Policy: Provides Real Time Cloud Access to Dynamic Black Lists
NOTE: TDM gateway is
used for connection
to TDM services
Feature
Benefit
Strategic
Differentia-
tor
CUBE Clustering Deployment
Options with vCUSP.
Scalability and Flexibility of CUBE
deployments for signaling & media flows C
CUBE upgrade to RP3 on ASR 1K Scalability of CUBE deployments C
Support SIPREC v17 for standards
based media forking & recording
Support recording solutions with 3rd party
recording servers and call control
Enhance signaling with WEB-RTC
Mobile Advisor Solution
Improved WEB-RTC-based Contact Center
Solution E
CUBE Enhancements (Sneak Peek) in Next 18 Months
C
C
C
E
Extending CUBE Flexibility Enabling a Range of Architectures for SIP-based Services
• Dynamic vCUBE activation & deactivation • Enable dynamic activation of virtual SBC containers to achieve “unlimited scaling”
• Separation of CUBE Signaling from CUBE Media Control • Establish concept of Media Gateway for transcoding, forking, conferencing, etc.
• Enhanced high availability between Data Centers & Multiple CUBE Pairs • Enable CUBE Clustering, even across geographic boundaries.
CUBE Recording Solutions - Supporting the Finalized SIPREC RFC
5
8
• Call agent independent
• Configured on a per Dial-peer level to fork RTP
Cisco MediaSense
(authentication disabled w/o UCM)
Partner Application • CUBE sets up a stateful SIP
session with MediaSense server
• After SIP dialog established,
CUBE forks the RTP and sends
it for MediaSense to record
SIP SIP
SIP
SP SIP
CUBE
RTP
RTP RTP
MediaSense
Dial-peer based
A
WEB-RTC in Action: Cisco Mobile Advisor
• Add communications to mobile apps & web pages
• Includes: Voice, Video & Chat
• Products: Mobile Advisor Client SDK
• Includes In-app communications and real time assistance features:
• Screen Sharing
• Co-browsing, Annotation etc.
• Products: Mobile Advisor Client SDK and Palettes (for non SIP based Context)
In-App Communications
In-App Live Assist Mobile Self Service
• Map from legacy self-service apps (Visual IVR)
• Products: Palettes Visual IVR
Mobile Advisor – Foundation Products
Palettes Client SDK
• WebRTC signaling engine • HD voice (Opus) & HD video
(VP8/H.264) • Instant messaging & presence • Application event distribution (AED) • SIP registrar & authentication • VP8 --> H.264; G.729 --> G.711
transcoding • Media port multiplexing • Web Based Management • Interop: Cisco UCM, UCCE/X, CUPS,
CVP & MS Lync
Web Gateway / Media Broker
Web & Mobile Client App SDKs
• Javascript API (JSON/REST) • iOS & Android native app SDK • Browser support (Chrome, Firefox,
Opera) • Outgoing & incoming calls • Multiple line handling • Adjustable resolution • Authenticated session (web app ID) • Sample client code & applications
In-App Communications Mobile Self-Service & Context
• IVR bypass
• Visual IVR
• Web rendering (HTML5)
• iOS app rendering (Objective C)
• Android app rendering (Java)
• Sample client code & applications
Palettes Server
• Agent context relay
• Rules based XML manipulation
• Dynamic code generation
• VoiceXML rules handling
• Genesys T/I-server integration*
• Cisco UCCE/UCCX integration*
MA
Palettes
MA Web Gateway
& Media Broker
MA Client
SDKs
MA Foundation Products
• Provides general purpose SBC features in a variety of form factors
• As a router-integrated element
• As a stand alone appliance
• As a virtual platform
• Provides Greatest Flexibility for deployment of SIP Services
• Architectural choice
• Deployment choice
• Provides greatest Protection against telephony denial of service (TDoS)
Cisco Collaboration Edge Architecture: Summary Summary of Cube Advantages for Deployment of SIP-based Services
Mobile Workers
B2B
Third-Party (Including
TDM/IP PBX)
Analog Devices
PSTN or IP PSTN
Branch Office
Cloud Services
Teleworker
Headquarters
Consumers