SRTG Update
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Cisco VG350
High Density Analog Voice
Gateway
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Port Density
August 2012
VG350
High Density Analog Gateway
96 – 160 Ports
C3945 112-FXS Bundle
VG224
VG204
VG202
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Platform / RU
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SM-D-72FXS
72 Port FXS Double Wide
Service Module
VG350
High Density Analog Gateway
SM-D-48FXS-E
48 Port FXS OPX-Lite Double
Wide Service Module
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4 EHWIC Slots
Support additional FXS
and FXO VICs
PVDM3 DSP Modules
Latest and greatest DSP technology
Noise Cancelation
Acoustic Shock Prevention
Redundant Power Supply Option
3 GE Ports
2 Flash storage slots
Standard console and
aux ports
2 Double Wide Service Module Slots
High Density Analog DWSM Slots
Support
72 Port FXS DWSM
48 Port OPX-Lite FXS DWSM
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USB Port
Console over USB
Convenient Storage
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• Modified ISR 3945 chassis dedicated for Voice Gateway function
• Software feature parity with VG2xx
Voice, fax and modem
Comprehensive set of codecs
SIP, H.323, SCCP and MGCP protocols
TLS / SRTP
Supported with CUCM 7.1.5, 8.0.3, 8.5.1, 8.6.2 and 9.0.1
Supported with CUCME / SRST 7.1, 8.0, 8.1, 8.5, 8.6, 8.8, 9.0 and 9.1
• HWIC slots can be used for additional FXO and FXS ports
VIC2-xFXO, VIC3-xFXS/DID and VIC3-2FXS-E/DID (Long loop length)
• Supports Online Insertion and Removal (OIR) of DWSMs
• Redundant Power Supply option
• Lightening protection
• IOS Release: 15.2(4)M [PI19]
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Ports 24 – 47
Ports 46 & 47
connected to PSTN
FXO Bypass failover
ports
Ports 0 – 23
Ports 0 – 3 support
long loop length
Standard RJ21 Connectors
• SM-D-72-FXS
Ports 48 – 71
2 PSTN FXO Bypass Failover Ports
Port 46 FXO Bypass Port 0
Port 47 FXO Bypass Port 1
OUT FXO Port
IN PSTN
72 port FXS Double Wide Service Module
Loop length – 3,300 ft with 26 AWG
First 4 ports will support OPX-Lite long loop length
11,000 ft with 26 AWG
18,000 ft with 24 AWG
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Ports 24 – 47
Ports 46 & 47
connected to PSTN
FXO Bypass failover
ports
Ports 0 – 23
Standard RJ21 Connectors
• SM-D-48-FXS-E
2 PSTN FXO Bypass Failover Ports
Port 46 FXO Bypass Port 0
Port 47 FXO Bypass Port 1
OUT FXO Port
IN PSTN
48 port OPX-Lite FXS Double Wide Service Module
All ports will support long loop length
11,000ft with 26 AWG
18,000ft with 24 AWG
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Cisco Unified
Communications
Manager Cluster
Cisco
Unity Cxn
VG350 Analog
Gateway
A
VG350
Corporate
IP Network
T1
T1
CAS/PRI
VG350
FXO
VG350 Analog
Gateway
ISR 2900/3900
with SRST
PSTN
Optional
Redundant
Links
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SCCP / H.323 / SIP
T1
Corporate
IP Network
FXO
VG350
Cisco Unified
CME
PSTN
VG350 Analog
Gateway
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• Supported at DWSM, EHWIC and PVDM3 module levels
• Modules can be powered off at scheduled times to conserve energy
• During “power off” period, FXO Bypass Failover will still be able to
provide dial tone on ports 46 and 47
• CLI is consistent with ISR G2
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50%
savings in
Rack
Space
6x VG224 (6 RU)
1x VG350 (3 RU)
• 6 to 1 savings in
83%
savings in
Infrastructure
&
Management
Switch port
Power supply
Console connection
Configuration change
Software update
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• Plug-n-play
Learn CUCM IP address from DHCP response
Download configuration from CUCM and provision the ports for SCCP
• IOS CLI Enhancements
4 CLIs per dial-peer = 576 CLIs for 144 ports 2 CLIs
4 CLIs per voice-port = 576 CLIs for 144 ports 8 CLIs
1 CLI can configure all the ports in the system for SCCP
CLI to check if a phone is connected to a port
• Cisco Management Applications Support
Cisco Unified Provisioning Manager 9.0
Cisco Unified Operations Manager 9.0
BATS Tool
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Feature Set and Roadmap
Analog Voice Gateway
Q1 CY10
Q3 CY12
15.0(1)M
15.1(1)T
15.2(4)M
Features:
• SIP VMWI (DC Voltage,
FSK)
Q4 CY10
15.1(3)T
To Ship
Not Yet Committed
Q4 CY09
Features:
• Distinctive ringing for
SCCP
• Enhancements to FAC
support – parity with VG248
Shipping
Features:
• VG350 high density FXS
& OPX-Lite analog
gateway
Future
Features:
• VG224 refresh
• SIP line side
enhancements
• CLI enhancements for
manageability
Features:
• Secure SCCP with CUCM
• Configurable call waiting
tone cadence
• Support for Media
Renegotiation Call flows 3PCC
• SNR, cBarge – across
analog / IP phone
• Serviceability
enhancements
Note: Future dates and features are estimates–subject to change
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CUBE Enterprise Update
Collaboration VT
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Enterprise 1
Enterprise 2
IP
IP
CUBE
IP
SIP
SIP
CUBE
Rich Media (real time Voice, Video, Screenshare) Rich Media
SESSION
CONTROL
SECURITY
INTERWORKING
DEMARCATION
Call Admissions
Control
Ensuring QoS
Statistics and Billing
Redundancy/
Scalability
Encryption
Authentication
Registration
SIP Protection
Firewall Placement
Toll Fraud
SIP - SIP
H.323 - SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fault Isolation
Topology Hiding
Network Borders
L5/L7 Protocol
Demarcation
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50-150
ASR 1004/6 RP2
20-35
Calls Per Second
3900E ISR G2
17
ASR 1002
3900 ISR G2
8-12
2801 ISR
<5
2900 ISR G2
ISR 3945 Transcoding of ~800 calls
800/1861 ISR
4
<50
ASR1001
Highest density
SBC:
10,000 sessions
in 1RU
ASR1006
Highest SBC
transcoding
capacity: 9,000
G729 to G711 calls
600-800
900-1000
1500-1700
2000-2500
10-12K
12-16K+
Active Voice Call (Session) Capacity
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ISR
ISR G2
CUBE
ASR
Parity
with ISR
CUBE
Vers.
IOS XE
Release
FCS
ASR
CUBE
Vers.
2800/ 3800
2900/
3900
FCS
1.3
12.4.22YB,
15.0.1M
15.0.1M
Oct 2009
<50%
1.3
15.0(1)S3
July 2010
1.4
15.0.1XA,
15.1.1T
15.1.1T
Apr 2010
<50%
1.4
15.0(1)S4
July 2010
8.5
15.1.2T
15.1.2T
July 2010
<50%
1.4
15.1(1)S
Nov 2010
8.6
15.1.3T
15.1.3T
Nov 2010
<50%
1.4
15.1(2)S2
Mar 2011
8.7
15.1.4M
15.1.4M
Apr 2011
~50%
1.4
15.1(3)S3
July 2011
8.8
EoL
15.2.1T
July 2011
~70%
1.5
15.2(1)S1
Nov 2011
9.0
EoL
15.2.2T
Nov 2011
NA
NA
NA
9.1
EoL
15.2.3T
Mar 2011
NA
NA
NA
NA
EoL
NA
NA
>75%
1.6
15.2(2)S
Mar 2012
NA
EoL
NA
NA
>85%
1.7
15.2(4)S
July 2012
9.2
EoL
15.2.4M
Oct 2012
>90%
9.2
15.3(1)S
Nov 2012
9.3
EoL
15.3.1
Mar 2013
>90%
9.3
15.3(2)S
Mar 2013
Internal: http://zed.cisco.com/confluence/display/VOICE/IOS+SBC+Release+Information
Cisco.com: Table 2 in the CUBE FAQ at www.cisco.com/go/cube
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• 15.1.2T added support for media preservation only
• This release adds support for signaling preservation – signaling
based disconnect and other supplementary services will work postswitchover
• Also added stateful failover for Software MTP based calls on router
Call Handling
X
CUBE
Active
Active calls are preserved
via checkpointing at the
time of failover
SIP SP
Standby -> Active
CUBE
New calls set up after the failure go through the
Standby (now Active) CUBE
CUBE ISR G2 High Availability (HA) Using HSRP Configuration Example
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a0080b40d82.shtml
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• New CLI mechanism for control of DSP for transcoding
G.729 30 ms
• Supported only with CUBE invoked DSP for transcoding
SP VoIP
• LTI Supported on both ISR and ASR platform
DSPs must be local to CUBE
Transcoding
Xcod
DSPs using LTI
• BENEFIT:
• Simplified CLI configuration for transcoding
dspfarm profile 2 transcode universal
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application CUBE
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CUBE
Cisco Unified
Border Element
Enterprise
VOIP
IP Phones:
G.711, G.729 20 ms,
G.722
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The Per Interface Port Management feature provides the capability where the port
range is managed per IP address range.
CUBE can be configured with multiple IP address ranges, where each range shares
the same port ranges with a maximum port range of 16K to 32K.
BENEFITS:
• INTEROPERABILITY - some third party applications use specific RTP port
ranges. This feature allows CUBE to conform to those applications.
• SECURITY CONFIGURATION - Firewall can be given more precise range of ports
that will be used for RTP traffic.
Example
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voice service voip
media-address range 8.8.8.1 8.8.8.2
media-address range 9.42.30.45 9.42.30.46
rtp-port range 20000 - 30000
IP Address Range
RTP Port Range
8.8.8.1 – 8.8.8.2
20000-30000
9.42.30.45 – 9.42.30.46
20000-30000
Global default (all other IP addresses)
20000-30000
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• PVDM3 used to clean audio by removing loud and distracting
background noise
• Enhances user experience by delivering quality audio
• Supported on TDM-IP, CUBE and Conference calls
• Enabled globally on router or selectively per dial peer basis
Gateway detects loud
background noise and
“cleans” media
Background noise
generated on a call
from a coffee house
Unified CM
A
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• Sudden loud high pitch feedback tone, a.k.a acoustic shock can a
workplace safety concern especially for call center workers.
• PVDM3 DSPs have the ability to detect acoustic shock neutralize its
effect
• Detects the loud, persistent, dominant single frequency tone in a
voice channel and performs corrective action
• Supported on TDM-IP, CUBE and Conference calls
Contact Center
CVP
IP
VXML
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Gateway Services API
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API Designed to:
• Allow users or policies to dynamically control
the gateways to the voice network
• Provide analytics of the voice network through
real time monitoring.
• Integrate into any web application using
WSDL-compatible development tools
• Allow 3rd parties to integrate applications onto
Cisco ISR using UCS express
• Simplify management and architecture
• Single platform using Cisco ISR
• Enterprise-wide solutions (TDM+SIP)
• Combine with data solutions
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Harassing / Threatening
Callers
Avoid productivity loss & safety
issues
Toll Fraud
Corporations lack real-time
defense
Voice Service Abuse &
Theft
Contact Center Fraud
& ID Theft
Ensure employee voice network
use matches business objectives
Legal risk and financial losses for
corporations and customers
Capacity Monitoring
Enables better network planning
and staffing requirements
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Unauthorized Fax or
Modem Usage
Most commonly found issue
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• NICE:
Using open APIs to dynamically control media recording
Eliminates need to record all voice calls: User-controlled or policy-controlled
Ideal for call centers
• MediaSense
Using open APIs to dynamically control media
• CUCM:
Enable IP-PBX to gain increased control of Gateway
• E-Loyalty / Teletech & Vodafone
Mobility applications
• WEBEX (Roadmap features)
Integrating with WebEx so end-users can turn on/off background noise reduction during
conference calls
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CME / SRST / CUE /
E-SRST 9.0 Update
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NEW
Cisco Unified Communications Manager Express 9.1
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• FCS: Q3 CY 2012
• Trivial mailbox PIN checks for end-users
• Unified Communications 9.0 inter-op
CUCM, CUC, CME, SRST 9.0
• 69xx & 89xx phone support for VoiceView
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New Features
• Support for Cisco Hosted Collaboration
Solution (HCS)
Customer Benefits
•Ensures telephony reliability for cloud
delivery
•Maintain business continuity during
outages
Partner Benefits
•Easier sales process – helps overcome
customer concerns about cloud reliability
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Product
Picture or
Generic
Graphic
Placeholder
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SRST
E-SRST
Ideal for customers who require:
• Basic phone features during failover
• Support of SIP & SCCP phones
• Supports Cisco Unified CM and Cisco
Unified CMBE
Ideal for customers who require:
• Full phone features during failover
• Hunt Group, Park Pickup, Speedials
• Simplified administration
• Automated provisioning & deployment
• Better security
• Supports Cisco Unified CM only
CUCM
Headquarters
SRST
Manager
Branch sites
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NEW
Enriching the Survivability Experience
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Headquarters
CUCM
1.
3.
IP WAN
A
4.
5.
Branch 1
SRST Manager
2.
PSTN
4.
5.
Branch n
Virtualised
1.
Provision the branch in CUCM
2.
Configure SRST Manager to connect to CUCM for provisioning info
3.
Provision branch router for IP connectivity and credentials
4.
SRST Manager sends provisioning information to each branch Router for Phone users ,
dialplan restrictions etc
5.
SRST Manager maintains any moves, adds and changes across HQ and branches
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New CCO Websites
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Thank you.