CUBE-EnT Preso For HCS Deployathon - V5
CUBE-EnT Preso For HCS Deployathon - V5
Cisco Confidential
Disclaimer
The Cisco products, service or features identified in this document may not yet be
available or may not be available in all areas and may be subject to change without
notice. Consult your local Cisco business contact for information on the products or
services available in your area. You can find additional information via Ciscos World
Wide Web server at http://www.cisco.com. Actual performance and environmental costs
of Cisco products will vary depending on individual customer configurations and
conditions.
Discussion Agenda
66%
IP PSTN
CUBE
IP
SIP
Enterprise 2
CUBE
IP
SIP
Session
Control
Call Admissions
Control
Trunk Routing
Ensuring QoS
Statistics and Billing
Redundancy (HA)
Scalability
Security
Interworking
Demarcation
Encryption
Authentication
Registration
SIP Protection
Voice Policy
Firewall Placement
Toll Fraud
SIP - SIP
H.323 - SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fault Isolation
Topology Hiding
Network Borders
L5/L7 Protocol
Demarcation
CUBE INTEROPERABILITY
Verified by
between IP networks
Demarcation
SIP TRUNK TO CUBE
Interworking
Session control
CUBE
Mobile
Cisco B2B
Security
Cisco SME centralizes
network control
Cisco Session
Management
IM, Presence,
Voicemail
Video
3rd Party IP
PBX
TDM PBX
Leverages
installed base
and knowledge
base
Integrated SBC
and TDM
Gateway
Simplifies
transition from
to IP PSTN
Broadest Scale of
price
performance
Voice
Security Policy
TDOS
protection with
granular policybased
enforcement
Integration with
Cisco
Collaboration
Cisco Unified
CM recording
solutions
CVP & UCCX
call center
solutions
WEBEX
integration for
Cloud Connect
Audio
Scalability
Unequaled scalability in price and performance
ASR 1006
& CUSP
ASR 1004
ASR 1006
50150
2035
Calls per
Second
17
3900 ISR
812
4300 ISR
NanoCUBE
2900 ISR
<5
Capacity: 3,0006,000
ASR 1001-X
ASR 1002-X
Capacity: 16,000
Capacity:
10,000 14,000
ASR +
virtualized CUSP
Capacity: 8002,500
Capacity: 1001000
Capacity: Up
to 50
<50
500600
1,000
2,500
6,000
12,000
16,000
64,000
NICE
MediaSense
RTP
SIP
SIP
RTP
SIP
SP SIP
SBC
CUBE
CVP
IVR
SIP
SBC
CUBE
ASR
Server
INSPECT - SCORE
Voice Security
Policy for TDOS
Mitigation
TEST
Inbound
Calls
A
No Cost
ISR-4K
Migration from
TDM Circuits to
SIP Trunks
PROBE
SP IP
Network
High Cost
TDM
Traditional
PSTN
SIP
H.323
CUBE
SIP
SBC
SP VOIP
Servivces
Infonetics Research
Market Analysis CY2014 Revenue
26%
35%
14%
13%
4Q15
Fcst
Note: There are at least 10 other Vendors not shown in
this Chart that would comprise OTHER
11%
Cisco
Oracle / Acme
Audio Codes
Other
Sonus
SP transition to SIP services worldwide is creating rapidly growing demand for SBC
CUBE is the Enterprise SBC leader (ranked #1 in global market share by INFONETICS/IHS from CY2013 thru CY2015)
CUBE is Used by 17,000+ organizations with 14 million+ trunk sessions in 160+ countries
CUBE offers outstanding interoperability with both SP SIP services and 3rd party IP-PBXs
TDM/VOIP integration
Call Center
Call Recording
WEBEX CCA
HCS
JP Morgan Chase
Chevron
Walmart
GM
FORD
GEICO
Apple
50150
2035
17
4400 ISR
3900 ISR
Capacity:
3,000-6,000
4300 ISR
812 NanoCUBE
2900 ISR
ASR 1001-X
ASR 1002-X
Calls per
Second
ASR 1004
ASR 1006
Capacity:
16,000
Capacity:
10,000
14,000
Capacity:
800-2,500
Capacity:
100-1000
Capacity:
100-600
<50
500600 1,000
2,500
6,000
12,000
16,000
https://cisco.box.com/SIPLabTheoryandAccess
https://cisco.box.com/CUBE
MPLS
Audio/
Video
SMB
Large
Enterprise
Internet
Audio
Only
Hosted
Services
SIP TRUNK
Call control on Customer Prem
MPLS
Mid-range
CUBE
SMB
Nano-CUBE
with 3PCC
CUBE
WEBEX CCA-SP
Internet
Multi-Tenant CUBE
for HCS
Hosted Services
SIP Line-side
Call control in the cloud
Large
Enterprise
CUBE
Large
Enterprise
IP PSTN
Aggregation SBC
Multi Tennant
High Session Capacity
Normalization
Policy
Services
SBC
Aggregation SBC
Multi Tennant
High Session Capacity
IP PSTN
SBC
HCS SBC
mTENANT CUBE-Ent
SBC
Normalization
Policy
Services (recording)
Call Center
CUBE
CUBE
HCS SBC
mTENANT CUBE-Ent
SBC
Normalization
Policy
Services (recording)
Call Center
CUBE
CUBE
IP PSTN
MPLS VPN
CUBE
nanoCUBE
Platforms
Business
Internet
C88X
C89X
SPIAD29XX
SIP trunk
Connection
MPLS VPN
LAN-I
SIP Lineside
Connection
Certified demarcation
CUBE
CUBE
Current nanoCUBE
3PCC Partners
Swisscom
Windstream
CLARO
IP-PBX
IP-Phone
LINESIDE SERVICE
Competitors
ORACLE
Edgewater
Adtran
AudioCodes
Requirements
WEBEX
CUBE
How
A
Enterprise
IP WAN
(MPLS)
CUBE
Headquarters
Benefit
Branch
Office
Branch
Office
Branch
Office
Distributed
IP PSTN
IP PSTN
Enterprise
IP WAN
Enterprise
IP WAN
CUBE
CUBE
Hybrid
CUBE
CUBE
CUBE
IP PSTN
Enterprise
IP WAN
CUBE
CUBE
CUBE
CUBE
CUBE
CUBE
SP
SP
Centralized
Centralized
Distributed
Distributed
Hybrid
Audio Only
1:1
Best
Good
Better
Audio Only
Multiparty
Good
Better
Better
Good
Better
Better
Worst
Best
Better
Worst
Best
Better
Collaboration Service
Hybrid
TDM PSTN
WEBEX
CUBE
How
A
Enterprise
IP WAN
(MPLS)
CUBE
Headquarters
Benefits
Branch
Office
Branch
Office
Branch
Office
CCA-SP - Topology
Customer
No Cisco UC
Requirement
SIP Trunk
Cisco WebEx
Collaboration Cloud
SP
SBC
WebEx
CUBE
SP Network
Leverage SP Network
TDM PRIs
SP Partner
Service Wrapper
Service Aggregation
Wholesale models
Cisco
HCS integration
Foundation architecture
for future network enabled
cloud services
WebEx Service
DMARC
Enterprise A
Network
Location 1
E1/T1, E3, T3
Ethernet, DSL
E1/T1, E3, T3
Ethernet, DSL
CUBE
PSTN
CUBE
CUBE
Enterprise B
Network
Location N
Location 1
Customer
WebEx
WebEx
(Conferencing
Platform)
On-net
PSTN GW
4
PSTN
Off-net
On Premise Equipment
SIP Trunks
Managed Services
- Service Set-up
- Managed Day 2 Support
WebEx Services
- Data Meetings
- CCA Ports
SIP
SIP
(dedicated)
PSTN/SIP
Network
Aggregation
Layer
CUBE-ENT
UC
Layer
(DC)
HCS WAN
HCS WAN
Customer A
Customer B
Customer Premise (CPE)
Customer C
Local Breakout
PSTN/SIP
Network
Introduction
Virtual Route Forwarding (VRF) is a technique which creates multiple virtual
independent networks within router
CUBE Enterprise Integrated with VRF enables a number of new use cases for
CUBE, including:
Logical separation of multiple CUBE call routing domains.
Enables multi-tenant deployments
Enablement of intra and inter VRF routing of voice and video calls
between VRF domains, without sharing IP addressing.
Security can be improved for calls between SP domains
Enablement of IP address Overlap with Multi VRF feature provide
seamless integration of new networks.
Allows greater flexibility in connecting to multiple SPs
Allows multiple legacy IP-PBXs with overlapping DN ranges to utilize
CUBE.
SP1
SP2
SP3
V
R
F
1
V
R
F
2
V
R
F
3
SP1
SP2
SP3
V
R
F
1
V
R
F
2
A
V
R
F
3
CUBE mTENANCY
Allowed
Required
Allowed
Not Allowed
Allowed,
(Requires use of Dial Peer
Group DPG CLI)
Allowed
Allowed
Allowed
IP Address Overlap
Allowed
Allowed
Not Allowed
Required
Yes
Yes
Listen sockets on UDP, TCP and TLS transports based on the VRF
Provision to configure RTP port ranges for each VRF and allocation
of Local RTP ports based upon VRF.
Ability to route the VoIP calls across different VRFs without need of
Route Leaks.
Multi-VRF
Basic Configuration
Inbound Dial Peer Match config
Inter / Intra VRF Call Routing config
Routing w/ Overlapped DN config
Routing w/ Overlapped IP config
Multi-Tenant
VRF
1
VRF
2
CUBE
ip vrf vrf1
rd 1:1
interface GigabitEthernet0/0/0
ip address 7.44.44.13 255.255.0.0
ip vrf forwarding vrf1
ip vrf vrf2
rd 2:2
interface GigabitEthernet0/0/1
ip address 6.44.44.13 255.255.0.0
ip vrf forwarding vrf2
VRF
2
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
interface GigabitEthernet0/0/0
interface GigabitEthernet0/0/1
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
interface GigabitEthernet0/0/1
interface GigabitEthernet0/0/0
dial-peer 22 preference 1
dial-peer 11 preference 1
VRF
2
Route based on
outbound
dial-peer group
ip vrf vrf2
rd 2:2
rd 1:1
interface GigabitEthernet0/0/1
interface GigabitEthernet0/0/0
Overlap local IP
dial-peer 22 preference 1
dial-peer 11 preference 1
VRF
2
CUBE
ip vrf vrf1
IP
Different remote IP
VRF
2
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
Interface GigabitEthernet0/0.1
encapsulation dot1Q 1 native
ip vrf forwarding vrf1
ip address 2.44.44.9 255.255.255.0
Overlap local IP
w/ VLAN subint
dial-peer 22 preference 1
dial-peer 11 preference 1
Interface GigabitEthernet0/0.2
encapsulation dot1Q 2
ip vrf forwarding vrf2
ip address 2.44.44.9 255.255.255.0
Different remote IP
CUBE
ip vrf vrf2
ip vrf vrf1
rd 2:2
rd 1:1
interface GigabitEthernet0/0/1
interface GigabitEthernet0/0/0
ip address 7.44.44.13 255.255.0.0
ip vrf forwarding vrf1
Overlap local IP
dial-peer 22 preference 1
dial-peer 11 preference 1
VRF
2
Overlap remote IP
VRF
2
CUBE
ip vrf vrf1
ip vrf vrf2
rd 2:2
rd 1:1
interface GigabitEthernet0/0/0
ip address 7.44.44.13 255.255.0.0
ip vrf forwarding vrf1
interface GigabitEthernet0/0/1
ip address 6.44.44.13 255.255.0.0
ip vrf forwarding vrf2
Available in phase 2
Inbound match based on VRF where SIP INVITE received.
For VRF 1, dial-peer 1 matched. For VRF 2, dial-peer 2.
E164 - aaaa
Registrar - 1
E164 - bbbb
Registrar - 2
Multi Tenant
Router#show run | sec tenant
Voice class tenant 1
registrar 1 ipv4:10.64.86.35:9051 expires 3600
credentials username aaaa password 7 06070E204D realm aaaa.com
outbound-proxy ipv4:10.64.86.35:9057
bind control source-interface GigabitEthernet0/0
Voice class tenant 2
registrar 1 ipv4:9.65.75.45:9052 expires 3600
credentials username bbbb password 7 110B1B0715 realm bbbb.com
outbound-proxy ipv4:10.64.86.40:9040
bind control source-interface GigabitEthernet0/1
E164 - aaaa
E164 - bbbb
Registrar - 1
OB-2 & Bind-2
Registrar - 2
Configurations
** (Showing only couple of configs as an example. All the configs under tenant be similar to configs under
sip-ua/voice service voip sip)
ip vrf vrf1
rd 1:1
interface GigabitEthernet0/0/0
ip address 7.44.44.13 255.255.0.0
ip vrf forwarding vrf1
VRF
2
CUBE
ip vrf vrf2
rd 2:2
interface GigabitEthernet0/0/1
ip address 6.44.44.13 255.255.0.0
ip vrf forwarding vrf2
Supported
Comments
MultiVRF / Multi-Tenant
support
Yes
No
Yes
NOW
Yes*
Suppression of UPDATE
Yes
Yes*
Yes
NOW
Yes
NOW
Setting of DSCP
precedence
Yes
NOW
Fixed BlackList
Yes
NOW
Supported
Comments
SIP Normalization
Yes
Now
Yes
Yes
Yes
No
InterVRF Routing
Yes
Yes
July 2014
Yes
July 2014
Dynamic Blacklist
Yes
Now
??
Under investigation
TBD
ASR1K
ISR-G2
4300/4400 (XE3.13.1)
vCUBE (XE3.15+)
Redundancy-Group
Infrastructure
HSRP Based
Redundancy-Group
Infrastructure
Redundancy-Group
Infrastructure
Not Available
Exists
Exists
Not Available
Media Forking
XE3.8
15.2.1T
XE3.10
Exists
XE3.6
Exists
Exists
Exists
DSP Card
SPA-DSP
PVDM2/PVDM3
PVDM4
Not Available
Not Available
Not Available
Transcoder Implementation
SCCP or
LTI (starting IOS 15.2.3T)
Exists
Exists
Exists
Exists
Web-based UC API
XE3.8
15.2.2T
Exists
Exists
Exists
15.2.3T
Exists
Not Available
XE3.9
15.3.2T
Exists
Not Available
Not Available
Exists
XE3.11
Not Available
VXML GW
Not Available
Exists
Not Available
Not Available
Calls Per
Second
(CPS)
CUBE
FlowThru
Calls
CUBE +
SW MTP
ASR1001-X
50
12000
ASR1002-X2
55
ASR1004 RP2
ASR1006 RP2
Platform
SIP TLS
w/SRTPRTP3
CUBE
Controlled
Recording
CUCM 10.X
Controlled
Recording
Xcoded
Calls
5000
357
11643
6600
6000
6000
14000
5000
1071
12929
6600
7000
7000
70
16000
8000
2499
13501
10800
8000
8000
70
16000
8000
3927
12073
10800
8000
8000
For Contact Center deployments, the max recommended capacity for CUBE ENT on an ASR1006 RP2 is no more than 4000
sessions for complex call flows based on 1000 SIP messages per second handled by the platform
Transcoding
Please reach
to on
theG711-G729r8
cs-cube@cisco.com
for capacity
planning
questions
prior to
customer
quotations indicate the
1.
is out
based
(High Complexity
Codec)
sessions.
The numbers
listed
for CUBE+Xcoding
number of transcoded sessions supported based on max DSP capacity as well as non-transcoded sessions. The total of both
should not exceed the CUBE Flow-Thru Calls for the overall
number
sessions
supported
mix
of some transcoded
and
CPS and
Sessionofcounts
listed
in (1) and in
(2)aare
independently
tested. Session
some non-transcoded sessions.
capacity (2) can be achieved at about half the CPS(1) listed here
2. ASR1002-X based on ESP10
ATT
Sprint
CUBE CLUSTERING
Scaling CUBE using CUSP for Load Balancing
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
Outgoing
CUBE
INVITE
sip:5551000@sip.com:5060
user=phone SIP/2.0
Incomi
ng
INVITE
sip:2222000020@9.13.24.6:5060
SIP/2.0
Outgoing
CUBE
INVITE
tel:2222000020
SIP/2.0
User-Agent: Cisco-SIPGateway/IOS-15.2.3.T
Diversion: <sip:3000@9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
...
m=audio 6001 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
...
Configure
SIP Profiles
Apply to
Dial-peer or
Globally
For Your
Reference
User-Agent: SP-SBC
Diversion: <sip:9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
...
m=audio 6001 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
...
Configure Inbound
SIP Profile to add a
dummy user part
Apply to Dial-peer
or Globally
For Your
Reference
WORD
WWW-Authenticate
Remove update from allow header: -- USE THIS IF YOU DON'T WANT THE PEER SIP
EQUIPMENT TO NOT SEND A MID-CALL SIP UPDATE TO THE CUBE.
voice class sip-profiles 1
request ANY sip-header Allow-Header modify "UPDATE, " ""
response ANY sip-header Allow-Header modify "UPDATE, " ""
2.
Change the host part of the Remote-Party-ID header to @sp.com instead of @CUBE's-IPaddress: -- USE THIS IF YOU WANT TO CHANGE THE DOMAIN PART OF RPID HEADER TO
REFLECT THE SP's DOMAIN. ESPECIALLY TRUE WITH SP's WHO HAVE BROADSOFT AS
THEIR SIP CALL-AGENT
voice class sip-profiles 1
request ANY sip-header Remote-Party-ID modify @9.13.24.5 @SP.com
response ANY sip-header Remote-Party-ID modify @9.13.24.5 @SP.com
Change sendonly to inactive in reINVITEs, 200 Oks and ACK w/SDPs -- USE THIS IF YOU HAVE ONEWAY OR NO-WAY VOICE PATH ISSUES WITH 3rd PARTY SIP EQUIPMENT, ESPECIALLY WHEN THE
SIP PEER DOES NOT COME OUT OF SENDONLY / RECVONLY / INACTIVE STATE AFTER CALL IS
PUT ON HOLD
voice class sip-profiles 1
request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
response 200 sdp-header Audio-Attribute modify "sendonly" "sendrecv"
Apply this to the voip dial-peer towards the SP using "voice-class sip-profiles 1".
The "inactive" attribute can be in ACK message or in a response. To cover all possibilities, this profile
below will convert any/all inactive's / sendonly's to sendrecv.
voice class sip-profiles 1
request any sdp-header Audio-Attribute modify "inactive" "sendrecv"
request any sdp-header Audio-Attribute modify "sendonly" "sendrecv"
response any sdp-header Audio-Attribute modify "inactive" "sendrecv"
response any sdp-header Audio-Attribute modify "sendonly" "sendrecv"
Remove Referred-By from outgoing REFER message: -- USE THIS IF YOU WANT TO SIMULATE
VARIOUS CONDITIONS WHILE REPRODUCING REFER RELATED CALL FLOW ISSUES
voice class sip-profiles 1
request REFER sip-header Referred-By remove
5.
Remove double c-lines from IOS SDP: -- USE THIS IF YOU ENCOUNTER INTEROP ISSUES
RELATED TO IOS SENDING SESSION AND MEDIA LEVEL c= LINES IN SDP
If having two c-lines in the SDP causes problems you can filter out whatever level c-line.
The excerpt below is for removing the media level c-line:
voice class sip-profiles 1
request ANY sdp-header Audio-Connection-Info remove
response ANY sdp-header Audio-Connection-Info remove
6.
Add user=phone to the P-Asserted-Identity URI: -- USE IF YOU NEED TO ADD user=phone
PARAMETER ADDED TO THE PAID HEADER.
voice class sip-profile 1
request INVITE sip-header P-Asserted-Identity modify "sip:@" "sip:"
8.
Adding Subscription State: Active to NOTIFY messages for MWI: --- USE IF THERE IS AN INTEROP ISSUE
RELATED TO MWI NOTIFY
voice class sip-profiles 1
request NOTIFY sip-header Subscription-State add "Subscription State: Active"
"Subscription State: Active"
<---- if there is
Add "user=phone" to Req URI -- USE IF SP or PEER WANTS user=phone TO APPEAR IN THE Req URI of SIP
INVITEs
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
Convert "sip" to "tel" URI: -- USE IF YOUR SP NEEDS IOS TO SEND tel: URI INSTEAD OF sip: URI
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "sip .*)@[^ ]+" "tel:\1"
request INVITE sip-header From modify "<sip .)@.>" "<tel:\1>"
request INVITE sip-header To modify "<sip .)@.>" "<tel:\1>
10.
Convert c=0.0.0.0 to c=<valid IP> in SDP of ReINVITEs: -- USE IF YOUR SP (ex. ALCATEL PBX)
DOESN'T SUPPORT c=0.0.0.0 in SDP in SETUPS like CUCM -- CUBE -- ALCATEL PBX
voice class sip-profiles 100
request ReINVITE sdp-header connection-Info modify "0.0.0.0" "10.199.71.100"
request ReINVITE sdp-header Audio-connection-Info modify "0.0.0.0" "10.199.71.100
11.
Add a tag, such as a trunk group indication, in the From header of the INVITE: -- USE THIS IF YOU
WANT TO DISTINGUISH CALLS AS BELONGING TO DIFFERENT TRUNK GROUPS - THE TRUNK
GROUP IS DEFINED BY HITTING A DIAL-PEER WITH A UNIQUE TRUNK-GROUP NUMBER ADDED
BY THE PROFILE ATTACHED TO THAT DIAL-PEER
Disable SIP session timer: -- USE TO AVOID YOUR SP (ex. SONUS PBX) REQUESTING FOR
SESSION TIMER WHEN THE OTHER END (ex. MITEL) DOESN'T SUPPORT SESSION TIMER. MITEL
-- CUBE -- SONUS
voice class sip-profiles 100
request INVITE sip-header Supported modify "timer," ""
request REINVITE sip-header Supported modify "timer," ""
request UPDATE sip-header Supported modify "timer," ""
request INVITE sip-header Min-SE remove
request REINVITE sip-header Min-SE remove
request UPDATE sip-header Min-SE remove
13.
Suppress the advertisement of the INFO message: -- USE IF YOUR SP (e.g. Acme SBC) RESPONDS
WITH AN INFO MESSAGE THAT MAKES THE CALL FLOW FAIL
voice class sip-profiles 1
request ANY sip-header Allow-Header modify "INFO, " ""
response ANY sip-header Allow-Header modify "INFO, " ""
Outbound Calls
IP PSTN
CUBE
Inbound Calls
(919)200-2000
Site B
(510)100-1000
Site C
(408)100-1000
G729 Sites
SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
G711 Sites
(919)200-2000
Site B
(510)100-1000
Site C
(408)100-1000
G729 Sites
SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
G711 Sites
CUBE
Enterprise
abc.com
SBC
Enterprise
xyz.com
For Your
Reference
CUBE
INVITE sip:1234@cisco.com
dial-peer voice 200 voip
destination uri 1
session protocol sipv2
session target ipv4:10.1.1.1
voice-class sip requri-passing
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
INVITE sip:hussain@cisco.com
dial-peer voice 100 voip
incoming uri request 1
session protocol sipv2
voice-class sip call-route url
For Your
Reference
CUBE
INVITE sip:hussain@10.1.1.1
dial-peer voice 200 voip
destination uri 1
session protocol sipv2
session target ipv4:10.1.1.1
voice class uri 1 sip
host cisco.com
INVITE sip:cisco.com
For Your
Reference
CUBE
INVITE sip:cisco.com
dial-peer voice 200 voip
destination uri 1
session protocol sipv2
session target ipv4:10.1.1.1
voice-class sip requri-passing
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
REFER and 302, both consume and pass-through cases supported as well
CUBE
destination uri 1
Facebook Video
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.
Total Calls,
CPU, Memory
CUBE
Call Spike
Detection
CUBE
Max Bandwidth
based
Call #3 Rejected
by CUBE
Call #1 80Kbps
Call #2 80 Kbps
Call #2
Call #3
Call #3
Rejected by
CUBE
CUBE
Call #3 80 Kbps
CUBE
Total Calls,
CPU, Memory
CUBE
Call Spike
Detection
CUBE
Max Bandwidth
based
Call #3 Rejected
by CUBE
Call #1 80Kbps
Call #2 80 Kbps
Call #2
Call #3
Call #3
Rejected by
CUBE
CUBE
Call #3 80 Kbps
CUBE
CVP
G.711
G.729 /
G.711
SP SIP
SIP
CUBE
4
G.729
G.729
CVP
Transcoder Inserted
G.711
G.729 /
G.711
SP SIP
SIP
CUBE
4
G.729
G.729
Transcoder Dropped
REFER Consumption
A
3. INVITE
SIP SP
CUBE
2. INVITE
CVP
1. REFER
SIP SP
CUBE
2. REFER
CVP
1. REFER
refer consume
Configured globally or
at inbound dial-peer
Yes (default)
No (default)
REFER Pass-through
Yes (default)
Yes
REFER Consume
No
No (default)
REFER Consume
No
Yes
REFER Consume
Outcome
SIP Dialer
Sent:
Received:
INVITE sip:2776677@9.41.35.205:5060
SIP/2.0
UPDATE
sip:sipp@9.42.30.151:7988;transport=UDP
SIP/2.0
Via: SIP/2.0/UDP
SIP/2.0/UDP 9.41.35.205:5060;branch=z9hG4bK6F26CF
9.42.30.151:7988;branch=z9hG4bK-16368-1-0
Via:
..
.
event=detected
--uniqueBoundary
status=Asm
Content-Type: application/x-cisco-cpa
pickupT=2140
Content-Disposition: signal;handling=optional
maxActGlitchT=70
numActGlitch=12
Events=FT,Asm,AsmT,Sit
valSpeechT=410
CPAMinSilencePeriod=608
maxPSSGlitchT=40
CPAAnalysisPeriod=2500
numPSSGlitch=1
CPAMaxTimeAnalysis=3000
silenceP=290
CPAMaxTermToneAnalysis=15000
termToneDetT=0
CPAMinValidSpeechTime=112
noiseTH=1000
actTh=32000
SIP SP
CVP
Contact Center
CUBE
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
SP IP Network
CUBE
Enterprise Network
CUBE
CUBE
CUBE
Distributed
Recording
Distributed
Recording
CUCM
Centralized
Recording
Enhanced control CUCM has policy control over media forking on CUBE & GWs.
Better bandwidth utilization use any CUCM, gain selectivity in call forking.
Flexibility distributed or centralized architecture, use any vendors media recording.
Improved compliance - record even network-connected mobile devices.
Partner Application
Cisco MediaSense
(authentication disabled w/o UCM)
MediaSense
SIP
RTP
SIP
SIP
SP SIP
RTP
CUBE
RTP
Tenant-A
Tenant-B
MPLS
CUBE
ASA-FW-A
ASA-FW-B
N7k
PSTN CPE
Breakout
10GB link
CUBE
ASR
1000
Cube-Ent-A
(Act/Stby)
CUBE
1GB link
PSTN
Provider
FI-62xx
1GB link
N100V
UCS
Chassis
UCAPPS-Tenant-A
UCAPPS-Tenant-B
ASR
1000
Cube-Ent-B
(Act/Stby)
UC-App Customer-A
(Vlan inside)
(Default GW = HSRP-cust-a-inside)
vlan-cust-A(inside)
Mgmnt Vlan
ASA-FW
NAT
vlan-custA(inside)
FW-Context
CustA
vlan-custA(outside)
vlan-cust-A(inside)
vlan-Cust-A(inside)
HSRP-Cust-a-inside
Static route to ASA for all traffic out of
vlan
Cust-A-Outside-VRF
vlan-cust-A(outside)
vlan-Cust-A(outside)
Hsrp-Cust-a-outside
Static route to AS for UC Apps
Static route to CUBE-Ent for PSTN
Static route to ASA for Mgmnt
To Agg-B(Peer)
PE
PE
CE
Customer
Premise
MPLS Core
CUBE-Ent
Cust-A-VRF
vlan-custA(outside)
SBC
(sbc inside
adjacency/i
nterface)
2 HSRPs
2 Vlans
4 Static routes
2 VRF
3 BGP peers
Network
Network
ASR
1000
customers
Network
Keep alives
ASR
1000
ASR
1000
customers
Dual Chassis
Network
ASR
1000
customers
No Redundancy
Network
ASR
1000
customers
Control link used to communicate the status of the routers, data link used to transfer stateful
information from the SBC.
One interface used for PSTN facing SIP and RTP traffic
One interface used for Customer facing SIP and RTP traffic
Dial
peer
Dial
peer
Dial
peer