CUBE Product Update - cisco.com Software Release Mapping . Feature Benefit Strategic Differentia-tor Enhanced Call Routing Simplified and enhanced dial peers for advanced call routing

John Vickroy

Product Manager

Presented on July 30, 2015

CUBE Product Update

John Vickroy

Product Line Manager

Presented on July 30, 2015

CUBE & CUSP Product Update

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

http://www.level3.com/index.html

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

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

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.


destination e164-pattern-map 100

codec g729r8

session target ipv4:10.1.1.1


url flash:e164-pattern-map.cfg


destination e164-pattern-map 100

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

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


e164 919200200.

e164 510100100.

e164 408100100.


description Inbound DP via Calling

incoming calling e164-pattern-map 100

codec g729r8


url flash:e164-pattern-map.cfg


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



!

!


description DP Source SJ w/voice class dpg

incoming called-number 13..

destination dpg 10000


description DPG 10000

destination-pattern 1341



!



destination-pattern 13..



!



destination-pattern 134.



!



destination-pattern 1...



!

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]


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] CUBE


incoming uri request 1 dial-peer voice 200 voip

session protocol sipv2 destination uri 1

voice-class sip call-route url session protocol sipv2


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









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






voice-class sip requri-passing


host cisco.com

For Your

Reference

sip:10.1.1.1

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







session target sip-uri


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

http://www.comverse.com/

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


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

|-----------------------------------------------


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}



!

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

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 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

Prime Collaboration CUBE Performance metrics

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

CUBE Role in Mobile Advisor Architecture

• 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

CUBE Product Update - cisco.com Software Release Mapping . Feature Benefit Strategic Differentia-tor Enhanced Call Routing Simplified and enhanced dial peers for advanced call routing

Documents