Transcript
CUBE Product Update Presented on July 30, 2015 John Vickroy Product Manager
CUBE & CUSP Product Update Presented on July 30, 2015 John Vickroy Product Line Manager
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)
CUBE Adoption Market Statistics •
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)
Estimated Market Growth in SIP Sessions Worldwide per Infonetics 2014
CUBE Interoperability Proven Interoperability and Interworking with Service Providers Worldwide
Validated with service providers world-wide
Tested with 3rd party PBXs
Standards based
Cisco Interoperability Portal: www.cisco.com/go/interoperability
Cisco Unified Border Element (CUBE) Router-based or virtual Session Border Control (SBC) Enterprise 1
CUBE
IP
Rich Media
SESSION CONTROL
Call Admissions Control Trunk Routing Ensuring QoS Statistics and Billing Redundancy/ Scalability
SIP
IP
Enterprise 2
CUBE
SIP
IP
CUBE
(Real time Voice & Video)
Rich Media
SECURITY
INTERWORKING
DEMARCATION
Encryption Authentication Registration SIP Protection Voice Policy Firewall Placement Toll Fraud
SIP - SIP H.323 - SIP SIP Normalization DTMF/ PT Interworking Transcoding Codec Filtering
Fault Isolation Topology Hiding Network Borders L5/L7 Protocol Demarcation
CUBE: Primary Strategic Differentiators Unmatched by competitors’ SBCs A
B
C
D
E
SBC Integration on the Router
Integrated SBC and TDM Gateway
Broadest Scale of price performance
Voice Security Policy
Integration with Cisco Collaboration
• Solutions for the smallest to largest deployments
• TDOS protection with granular policybased enforcement
• Leverages installed base and knowledge base
• Simplifies transition from to IP PSTN
Enables flexible deployment models: centralized, distributed, hybrid
• Cisco Unified CM recording solutions • CVP call center solutions • WEBEX integration for Cloud Connect Audio
CUBE: Platform Portfolio Unequaled Price / Performance SBC Scalability ASR 1002-X Introduced in July 2012
50–150
ASR 1001-X Introduced in March 2014
20–35
4400 ISR Introduced in July 2013
Calls per Second
17
4300 ISR Introduced in Sept 2014
1001-X & 1002-X ASR
4400 ISR 3000-6000
8–12 NanoCUBE
3900 ISR 800-2,500
Oct 2013
ASR 1004/6 RP2
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 <5
800 ISR Up to 50
4
2900 ISR 100-600
<50
500–600
1,000
2,500
6,000
Active Voice Call (Session) Capacity
12,000
16,000
64,000
CUBE Market Leading Scalability Only SBC Platform to Extend Across All Customer Market Segments
SMB <100 sessions
Mid-Market Commercial
Enterprise >3000 Sessions
100 to 3000 sessions
CISCO CUBE
CISCO CUBE
CISCO CUBE
ISR 88X SPIAD 29XX
ISR-G2 ISR 43XX / 44XX
ASR 1006 ISR 43XX / 44XX ISR-G2
EdgeWater Adtran
Audio Codes InGate Avaya
ACME / Oracle Sonus
CUBE Market Segmentation Opportunity Ranking by Channel Customer Segment
Platforms
SIP service
Direct
Large Enterprise
Mostly ASR Some ISR
Trunk Lineside (remote worker)
XXX
ASR and ISR
Trunk and Lineside
XXX
Mostly ISR Some ASR
Trunk Lineside (HCS)
XXX
ISR
Trunk Lineside (HCS)
Cisco 8XX
Trunk Lineside (Broadsoft)
DoD Small Enterprise Large Commercial SMB
VAR
SP XX
XXX
XXX
XXX
XXX
XX
CUBE Architecture Flexibility Efficiently Supports All SIP Architectures for Voice or Video Services Centralized SIP Architecture
Distributed SIP Architecture
IP PSTN
IP PSTN
Enterprise IP WAN
Enterprise IP WAN
CUBE
CUBE
CUBE
CUBE
CUBE
Hybrid SIP Architecture IP PSTN Enterprise IP WAN CUBE
CUBE CUBE
CUBE
CUBE
CUBE
How to Choose a SIP Trunk Architecture Collaboration services should determine the architecture type Collaboration Service Audio only: 1 to 1 Audio only: multi-party conferencing
Audio & Video: 1 to 1 Audio & Video: multi-party conferencing Cloud Collaboration
Centralized Best Good Good Worst Worst
Distributed Good Better Better Best Best
Hybrid Better Better Better Better Better
CUBE enables WEBEX Cloud Connect Audio (CCA) Practical Application of Distributed SIP Trunking 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 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.
Headquarters
CUBE
Branch Office
CUBE
Branch Office
Benefit CUBE
Branch Office
•
Dramatic savings thru elimination of TDM, plus excellent conference experience thru efficient network usage.
CUSP: Optimization of SIP Signaling Simplify, Normalize and Balance Call Signaling between all SIP Network Elements IP-PSTN
IP-PSTN
CUBE
CUBE
CUBE
VXML
VXML
CUCM UCCE/X
CUBE
CVP
Without CUSP SIP Proxy
CUCM UCCE/X CVP
With CUSP SIP Proxy
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
CUBE CLUSTERING - Phase 1 Dynamic Load Balancing of SBC Resources for Highly Scalable SIP Gateway
USE CASE 2: CUSP for Internal Call Center Integration with UCCE / CVP
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
CUBE Software Release Mapping ISR G2
ASR / ISR-4K*/vCUBE (CSR)*
CUBE Vers.
2900/ 3900
FCS
CUBE Ent ASR Parity with ISR
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
11.1.0
3.16
15.5(3)S
July 2015
11.5.0
15.6(1)T
Nov 2015
11.5.0
3.17
15.6(1)S
Nov 2015
12.0.0
15.6(2)T
Mar 2016
12.0.0
3.18
15.6(2)S
Mar 2016
12.1.0
15.6(3)M
July 2016
>95% >95% >95% >95%
12.1.0
3.19
15.6(3)S
July 2016
* IOS-XE3.13.1 or later recommended for all ISR-4K series and XE3.15 for vCUBE 20
CUBE Vers.
IOS XE Release
FCS
CUBE Enhancements from Recent Releases Feature
Benefit
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
Strategic Differentiator
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 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 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. dial-peer voice 1 voip destination e164-pattern-map 100 codec g729r8 session target ipv4:10.1.1.1
A
SIP Trunk
Provides the ability to combine multiple destination-patterns targeted to the same destination to be grouped into a single dial-peer
SP SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
G711 Sites
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-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
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
A
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 SP SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
G711 Sites
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-patternmap.cfg 9192002010 5101001010 4081001010
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
”.
•
The inbound VOIP dial-peer references the dial peer group with a new CLI: “destination dpg ”
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.
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 I incoming called-number 13.. N destination dpg 10000
Incoming Dialed # 1341
1. Incoming Dial-peer is first matched
B O U N D
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 2. Now the DPG associated destination-pattern 1... session protocol sipv2 with the INBOUND DP is session target ipv4:10.1.1.4 selected !
O U T B O U N D
URI Based Dialing Overview INVITE sip:[email protected] INVITE sip:[email protected]
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. [email protected], [email protected], two different dial-peers are configured with the respective session targets
URI Based Dialing Enhancement – URI Pass Through INVITE sip:[email protected]
For Your Reference
CUBE
INVITE sip:[email protected]
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
•
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
URI Based Dialing Enhancement – ‘User’ portion non-E164 format
INVITE sip:[email protected]
For Your Reference
CUBE
INVITE sip:[email protected]
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
•
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
URI Based Dialing Enhancement – ‘User’ portion absent
INVITE sip:cisco.com
For Your Reference
CUBE
INVITE sip:cisco.com
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
•
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
URI Based Dialing Enhancement – Deriving Target host from Incoming INVITE Req-URI INVITE sip:[email protected]
CUBE
INVITE sip:[email protected] Skype
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
Facebook Video
session protocol sipv2 session target sip-uri voice class uri 1 sip
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.
NanoCUBE Deployment Scenarios Service Provider Call Control
NanoCUBE Hosted Service Small Business SIP Trunking Small Business
CPE
NANOCUBE 8xx
SIP
NANO -CUBE
SIP
SIP IAD
8xx
CUBE
CUCM
SIP
SIP
PRI TDM PBX
IP PBX
Enterprise
Hosted Service Small Business
3
SIP Trunking Small Business
PRI To SIP
CUBE Value Proposition to SP’s: Breadth of Scalability increases SP market size and reduces operation costs Managed Services
SIP TRUNK Call control on Customer Prem
Mid-range SBC
SMB
Low end SBC’s
CUBE
Hosted Services
SIP Line-side Call control in the cloud
High Capacity SBC
Large Enterprise
Dynamic CUCM Triggered Call Recording
Delivering Industry’s 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, uses any vendors media recording Improved Compliance - record even network-connected mobile devices
Debugging Made Easier 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. 2. 3. 4.
Critical Notifications Informational Verbose
Router# debug ccsip level 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 Categorize Debugs based on Functionality CUBE# show cube debug category codes 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 Categorize Debugs based on Functionality Categorization based on Functionality 1. 2. 3. 4. 5. 6. 7.
Audio/video/sdp/control Configuration /sip-transport CAC DTMF/FAX/Line-side Registration Sdp - passthrough Sip-profile/SRTP/transcoder
Router# debug ccsip feature < audio | cac | config | control | dtmf | fax | line | misc | misc-features | parse | registration | sdpnegotiation | sdp-passthrough | sip-profiles | sip-transport | srtp | supplementaryservices | transcoder | video >
Example: enabling DTMF and audio debugs only with default log level is considered. DTMF(32) debug code 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)
Audio(2) debug code
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
CUBE Enhancements Being Released in April Feature
Benefit
Strategic Differentiator
Virtualization of both CUBE and CUSP
Enable even greater flexibility in CUBE deployment models, including CUBE Clustering
C
SIP-based Call Progress Analysis
Allow Outbound Call Center over a SIP PSTN
E 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
E D
Enable use of voice policy even at the very highest scale of call connections ( >10,000 sessions)
D E
Voice Security Policy Performance Enhancements
Cisco Unified SIP Proxy (CUSP) A Critical Element of Cisco Collaboration Infrastructure Optimize, normalize, balance call signaling between all SIP elements IP-PSTN
IP-PSTN
CUBE
CUBE
CUBE
CUBE
VXML
VXML
CUCM UCCE/X
CVP
Without CUSP
CUCM UCCE/X CVP
With CUSP
Announcing CUBE & CUSP Virtualization Enhanced Deployment Flexibility and Scalability through Virtualization • 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
UCS Servers
Introducing vCUBE (CUBE on CSR 1000v) Virtual Architecture • CSR (Cloud Services Router) 1000v runs on a Hypervisor – IOS XE without the router
ESXi Container
ESP (data plane)
RP (control plane) Chassis Mgr.
IOS XE
Chassis Mgr.
QFP Client / Driver
Forwarding Mgr.
Forwarding Mgr.
FFP code
CUBE signaling
CUBE media processing
Kernel (incl. utilities) Virtual CPU
Flash / Disk
Memory
Console
Ethernet NICs
Mgmt ENET
CSR 1000 (virtual IOS XE)
vSwitch NIC
Hypervisor X86 Multi-Core CPU
Memory Banks
Hardware
GE
…
GE
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 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/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
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 coexistence
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 ASR 1002-X Introduced in July 2012
50–150
vCUBE to be added April 2015
20–35
4400 ISR Introduced in July 2013
Calls per Second
17
1006 ASR & CUSP
ASR 1001-X Introduced in March 2014
4300 ISR Introduced in Sept 2014
1001-X & 1002-X ASR
4400 ISR 3000-6000
8–12
3900 ISR 800-2,500
ASR 1004/6 RP2
ASR 1001 Highest Density 10,000 Session in 1 RU
4300 ISR 100-1000
ASR 1006
Highest Transcoding Capacity: 9,000 G729 to G711 Calls
ASR + virtualized CUSP 64,000 sessions enabled by CUBE clustering
<5
800 ISR Up to 50
4
2900 ISR 100-600
<50
500–600
1,000
2,500
6,000
Active Voice Call (Session) Capacity
12,000
16,000
64,000
•
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
Very Large Sessions 16,000
Integrates with CUBE to support CUBE clusters
Up to 64,000 calls
6000
•
Large Small to Medium
1
New Support for very large deployments with virtualized CUSP
64,000
CUBE: For Deployments of Every Size
Up to 16,000 calls
Up to 6000 calls
CUBE on ISR or virtual
CUBE on ASR CUBE on ASR or or virtual virtual with CUSP 9.0
Call Progress Analysis (CPA) on SIP Trunks Cisco Call Center supports an ALL SIP Environment (inbound and outbound!) Sent: Received: INVITE sip:[email protected]:5060 SIP/2.0 UPDATE sip:[email protected]: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-based CPA enables Cisco Outbound Call Center solution to support SIP trunk Connections through CUBE
SIP Dialer
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 / Voice Mail 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
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} !
For Your Reference
CUBE Monitoring 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
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
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
Loss, delay, jitter
CISCO-VOICE-DIAL-CONTROL-MIB, cvVoIPCallActiveTable
IP SLA
CISCO-RTTMON-RTP-MIB, rttMonJitterStatsTable , rttMonLatestJitterOperTable
Traffic Reports (Calls, Sessions, Capacity Planning, Errors)
Media Resources (DSPs)
Voice Quality
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
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
Prime Collaboration - Assurance CUBE Features Benefits matrix Features Monitoring Cisco Unified Border Element (CUBE)
Benefits
Detecting SIP trunk Outage
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. 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
Call Performance metrics
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. Additional CUBE KPIs such as call stats for deeper monitoring
Prime Collaboration CUBE Performance metrics
Telephony Denial of Service (TDoS) Dramatic Increase in number and severity of TDOS attacks 2014 TDoS Warnings: • • • •
FBI DHS NENA 911 APCO
Application Security (Malicious Calls)
Telephony DoS Social Engineering / ATO IRSF / Toll Fraud Robocalls / Scams / Vishing Harassing Calls UC federation
Network Security (Malicious Packets)
Rogue RTP Mal-Formed SIP Events Untrusted Sources Eavesdropping
CUBE: Per Call Security Policy Process Each and every call is filtered and evaluated for threat potential in real time TEST / PROBE
INSPECT - SCORE (All calls at no cost) 100%
• • • •
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)
• • • •
ENFORCE <5%
~20%
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
CUBE Voice Policy: Provides Real Time Cloud Access to Dynamic Black Lists Carrier TDM
Carrier SIP
CUBE
Landline Authentication
Mobile Authentication
NOTE: TDM gateway is used for connection to TDM services
Voice Policy
Location Authentication
Substantial Managed Blacklisting
Report, Record Reroute, Reject
OTHER DB’s SIP Quality and IP Anomalies
Assigned Numbers Valid Numbers Etc.
CUBE Enhancements (Sneak Peek) in Next 18 Months Feature
Benefit
Strategic Differentiator
CUBE Clustering Deployment Options with vCUSP.
Scalability and Flexibility of CUBE deployments for signaling & media flows
C C
CUBE upgrade to RP3 on ASR 1K
Scalability of CUBE deployments
C C
Support SIPREC v17 for standards based media forking & recording
Support recording solutions with 3rd party recording servers and call control
C
Enhance signaling with WEB-RTC Mobile Advisor Solution
Improved WEB-RTC-based Contact Center Solution
E 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
Dial-peer based
• CUBE sets up a stateful SIP session with MediaSense server
Partner Application
Cisco MediaSense (authentication disabled w/o UCM)
• After SIP dialog established, CUBE forks the RTP and sends it for MediaSense to record
MediaSense
SIP
A
RTP SIP
SIP SP SIP
RTP
• Call agent independent • Configured on a per Dial-peer level to fork RTP
CUBE
RTP
WEB-RTC in Action: Cisco Mobile Advisor In-App Communications
• Add communications to mobile apps & web pages • Includes: Voice, Video & Chat • Products: Mobile Advisor Client SDK
In-App Live Assist
• 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)
Mobile Self Service
• Map from legacy self-service apps (Visual IVR) • Products: Palettes Visual IVR
Mobile Advisor – Foundation Products Mobile Self-Service & Context
In-App Communications Web & Mobile Client App SDKs
Palettes Client SDK
• 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
Web Gateway / Media Broker • 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 • • • •
• • • • • •
MA Client SDKs MA Web Gateway & Media Broker
MA Palettes
MA Foundation Products
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*
CUBE Role in Mobile Advisor Architecture
Cisco Collaboration Edge Architecture: Summary Summary of Cube Advantages for Deployment of SIP-based Services •
Provides general purpose SBC features in a variety of form factors • • •
•
Provides Greatest Flexibility for deployment of SIP Services • •
•
As a router-integrated element As a stand alone appliance As a virtual platform
Architectural choice Deployment choice
Provides greatest Protection against telephony denial of service (TDoS)
Mobile Workers
Teleworker Headquarters
B2B
Branch Office
Consumers
Third-Party
PSTN or IP PSTN
(Including TDM/IP PBX)
Cloud Services
Analog Devices
