Cisco Live! :: Deploying SIP Trunks with Cisco Unified Border Element (CUBE/vCUBE) Enterprise | 2017

Deploying SIP Trunks with Cisco
Unified Border Element
(CUBE/vCUBE) Enterprise
Hussain Ali, CCIE# 38068 (Voice, Collaboration)
Technical Marketing Engineer
BRKUCC-2006

• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Serviceability
• Futures & Key Takeaways
Agenda

© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE (Enterprise) Product Portfolio
BRKUCC-2006 5
2900 Series
ISR-G2 (2901, 2911, 2921, 2951)
ASR 1004/6 RP2
Active Concurrent Voice Calls Capacity
CPS
<5
8-12
50-150
14-16K<50 500-600 900-1000
3900 Series ISR-G2 (3925, 3945)
17
3900E Series ISR-G2
(3925E, 3945E)
2000-2500
20-35
4
800 ISR
7000-10,000
50-100
12K-14K
ASR 1002-X
4500-6000
ISR 4451-X
ASR 1001-X
4000
ISR 4431
ISR-4K (4321, 4331)
ISR 4351
Introducing CUBE on
CSR
vCUBE [Performance
dependent on vCPU and
memory]
Note: SM-X-PVDM module
supported on XE3.16 or
later for ISR 4K platforms

Introducing IOS-XE Release 16
BRKUCC-2006 6
• New OS from the platform team with the intent of consolidating OS’ on different
product portfolio
• UX will be the same as IOS-XE, no difference to end user
• IOS-XE Release 16.3.1 support for UC (CUBE, CME, SRST)
Ø Impacts XE based (ASR1K, ISR4K, and vCUBE) platforms
Ø There will be no CUBE 11.5.1 for the XE based platforms [ASR1K, ISR4K,
vCUBE]. CUBE 11.5.2 (July 2016 release) will have newer and March 2016
features for the XE based platforms introduced in IOS-XE release 16.3.1
Ø IOS-XE 16 requires a minimum of ASR1001-X, 1002-X, 1004/1006 RP2,
ESP20 (Embedded Service Processor, SIP40 (SPA Interface processor)
Ø It will include all features up to and including IOS-XE 3.17 as well
• Due to new hardware requirements, customers will have the following migrations
options as IOS-XE 3.17 rebuilds will stop by June 2017
Ø Replace unsupported ASR1K hardware and upgrade to IOS-XE 16.3.1 or later
and continue to enjoy new feature set/support for any issues
Ø Drop using new feature set and move back to IOS-XE3.16 long maintenance
release for longer support

ISR G2 ASR 1K / ISR-4K/vCUBE (CSR)
CUBE Vers. 2900/ 3900 FCS CUBE Vers. IOS XE Release FCS
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
CUBE
Vers.
2900/ 3900 FCS
CUBE
Vers.
IOS XE Release 16 2 FCS
11.5.14 15.6(2)T14 Mar 2016 N/A 3 16.2.13 Mar 2016
11.5.2 15.6(3)M1 Dec 2016 11.5.23 16.3.2/16.4.13 Nov 2016
EOL EOL EOL 11.6.0 16.5.1 Mar 2017
CUBE Software Release Mapping
2
IOS-XE 16 requires a minimum of ASR1001-X, 1002-X, 1004/1006 RP2, ESP20 (Embedded Service Processor, SIP40 (SPA Interface
processor)
3
IOS-XE release 16.2.1 does not support CUBE functionality on the platforms. There is no CUBE version 11.5.1 for the XE based
platforms. All CUBE features from 11.5.0 (IOS-XE 3.17) and earlier versions along with CUBE 11.5.1 (March 2016 release) on ISR G2
are included in CUBE release 11.5.2 for the IOS-XE based platforms, IOS-XE release 16.3.1 [July 2016 release]
4
IOS 15.6(2)T will show CUBE Release version to be 12.0.0 but due to DDTS# CSCuz43735, rebuilds for this release train will align
to CUBE release 11.5.1, that is 15.6(2)T1/T2/T3/T4 and so on will be CUBE version 11.5.1
BRKUCC-2006

ISR G2 CUBE Ent
ASR Parity
with ISR
ASR 1K Series
CUBE Vers. 2900/ 3900 FCS CUBE Vers. IOS XE Release FCS
8.5 15.1(2)T July 2010 <50% 1.4 3.2 15.1(1)S Nov 2010
8.6 15.1(3)T Nov 2010 <50% 1.4.1 3.3 15.1(2)S March 2011
8.7 15.1(4)M April 2011 ~50% 1.4.2 3.4 15.1(3)S July 2011
8.8 15.2(1)T July 2011 ~70% 1.4.3 3.5 15.2(1)S Nov 2011
8.9 15.2(2)T Nov 2011 >80% 1.4.4 3.6 15.2(2)S Mar 2012
9.0
15.2(3)T/
15.2(4)M
Mar 2012 >85% 9.0 3.7 15.2(4)S July 2012
9.0.1 15.3(1)T 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
CUBE Software Release Mapping – Earlier Releases
BRKUCC-2006

ISR G2 CUBE Ent
ASR Parity
with ISR
ASR 1K / ISR-4K Series
CUBE
Vers.
2900/ 3900 FCS
CUBE
Vers.
IOS XE Release FCS
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
CUBE Software Release Mapping – Earlier Releases
BRKUCC-2006

Proven Interoperability and Interworking with
Service Providers Worldwide
Cisco Interoperability Portal:
www.cisco.com/go/interoperability
• Validated with Service
Providers World-Wide
• Independently Tested
with 3-Party PBXs in
tekVizion Labs
• Standards based
Verified by
10BRKUCC-2006
CUBE Interoperability

CUBE Call Processing
§ Actively involved in the call treatment,
signalling and media streams
§ SIP B2B User Agent
§ Signalling is terminated, interpreted and
re-originated
§ Provides full inspection of signalling, and
protection against malformed and malicious
packets
§ Media is handled in two different modes:
§ Media Flow-Through
§ Media Flow-Around
§ Digital Signal Processors (DSPs) are only
required for transcoding (calls with
dissimilar codecs)
13BRKUCC-2006
IP
CUBE
CUBE
IP
Media Flow-Around
§ Only Signaling is terminated on CUBE
§ Media bypasses the Cisco Unified Border
Element
Media Flow-Through
§ Signaling and media terminated by the Cisco
Unified Border Element
§ Transcoding and complete IP address hiding
require this model

Cisco Unified Border Element Basic Call Flow
1. Incoming VoIP setup message from originating endpoint
2. This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD,
DTMF method, protocol, etc.
3. Match the called number to outbound VoIP dial peer 2
4. Outgoing VoIP setup message
14BRKUCC-2006
Incoming VoIP Call Outgoing VoIP Call
dial-peer voice 1 voip
destination-pattern 1000
session protocol sipv2
session target ipv4:1.1.1.1
codec g711ulaw
codec g711ulaw
Originating
Endpoint -
1000
Terminating
Endpoint –
2000
CUBE
voice service voip
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip

CUBE
1000
1.1.1.1
2000
2.2.2.2
Incoming VoIP Call Leg
Matches an Incoming Dial-peer
Outgoing VoIP Call Leg
Matches an Outbound Dial-peer
INVITE /w SDP INVITE /w SDP
100 TRYING100 TRYING
VRF1 – 10.10.10.10 20.20.20.20 – VRF2
c= 1.1.1.1
m=audio abc RTP/AVP 0
c= 20.20.20.20
m=audio xxx RTP/AVP 0
180 RINGING180 RINGING
200 OK
200 OK c= 2.2.2.2
m=audio uvw RTP/AVP 0c= 10.10.10.10
m=audio xyz RTP/AVP 0
RTP (Audio)
ACK
ACK
1.1.1.1 10.10.10.10 20.20.20.20 2.2.2.2
Understanding the Call flow
BYE BYE
200 OK
200 OK
15BRKUCC-2006

CUBE Architecture
ISR G2 vs ASR1K vs ISR 4K vs vCUBE (CUBE on
CSR)

ASR/ISR-4K & ISR-G2 Architecture Comparison
§ ISR: Pkt fwd’ing and signaling are handled by the
same CPU
§ ASR: Pkt fwd’ing and signaling are handled by
different CPUs
‒ ESP must be programmed or instructed by the
control plane to do specific media functions
‒ Performed by Forwarding Plane Interface (FPI)
I/O ESP I/O
Kernel
IOS-XEIOS-XE
Msg I/f
Control Plane
Data (Forwarding) Plane
ASR/ISR-4K (IOS-XE) Architecture
RP
IOS
CPU
I/O I/O
Control
Plane
Data Plane
ISR G2 Architecture
Signaling
Signaling
Media
BRKCOL-2125 17

Introducing vCUBE (CUBE on CSR 1000v)
Architecture
• CSR (Cloud Services Router) 1000v runs on a Hypervisor – IOS
XE without the router
18BRKUCC-2006
Console Mgmt ENET Ethernet NICsFlash / DiskMemoryVirtual 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
CSR 1000v (virtual IOS-XE)

Introducing vCUBE (CUBE on CSR 1000v) –
Cont’d
• CSR1000v is a virtual machine, running on x86 server (no specialised hardware) with
physical resources are managed by hypervisor and shared among VMs
• Requires APPX (No TLS/SRTP) or AX (All vCUBE features) CSR licensing package to
access voice CLI and increase throughput from 100 kbps default. CUBE Licensing
follows ASR1K SKUs and still trust based
• No DSP based features (transcoding/inband-RFC2833 DTMF/ASP/NR) available
• Specs based hardware supported but performance benchmarked for Cisco UCS B and
C series only
• vCUBE Tested Reference Configurations [UCS base-M2-C460, C220-M3S, ESXi 5.1.0
& 5.5.0]. ESXi 6.0 supported with IOS-XE 16.3.1 or later
BRKUCC-2006 19

Step 1: Enable CUBE Application on Cisco Routers
voice service voip
mode border-element license capacity 20 à License count entered here not enforced though this CLI is
required to see “show cube” CLI output
allow-connections sip to sip à By default IOS/IOS-XE voice devices do not allow an incoming
VoIP leg to go out as VoIP
2. Configure any other global settings to meet SP’s requirements
voice service voip
media bulk-stats à To increment Rx/Tx counters on IOS-XE based platforms. W/O this CLI, it will show 0/0
sip
early-offer forced
header-passing
error-passthru
3. Create a trusted list of IP addresses to prevent toll-fraud
voice service voip
ip address trusted list à Applications initiating signalling towards CUBE, e.g. CUCM, CVP,
ipv4 66.77.37.2 ! ITSP SIP Trunk Service Provider’s SBC. IP Addresses from dial-peers with “session target
ipv4 10.10.1.20/28 ! CUCM ip” or Server Group are trusted by default and need not be populated here
sip
silent-discard untrusted à Default configuration starting XE 3.10.1 /15.3(3)M1 to mitigate TDoS Attack
1. Enable CUBE Application
BRKUCC-2006 21

Step 2: Configure Call Routing on CUBE
22BRKUCC-2006
IP PSTN
A
TDM PBX
SRST
CME
MPLS
Enterprise Branch
Offices
Enterprise
Campus
CUBE with High
Availability
Active
Standby
CUBE
CUBE
PSTN is now
used only for
emergency calls
over FXO lines
WAN Dial-PeersLAN Dial-Peers
• Dial-Peer – “static routing” table mapping phone numbers to interfaces or IP addresses
• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and
receiving calls to & from the PBX
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for
sending & receiving calls to & from the provider

WAN Dial-Peer Configuration
23BRKUCC-2006
description *** Inbound WAN side dial-peer ***
incoming called-number 408527….$
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
codec g711ulaw
dtmf-relay rtp-nte
no vad
Outbound Dial-Peer for call legs from CUBE to SP
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 91[2-9]..[2-9]......$
session target ipv4:<SIP_Trunk_IP_Address>
codec g711ulaw
dtmf-relay rtp-nte
no vad
Specific to your DID range
assigned by the SP
Dial-peer for making long distance
calls to SP, based on NANP (North
American Numbering Plan)
Note: Separate outgoing DP to be created for Local, International,
Emergency, Informational calls etc.
Inbound Dial-Peer for call legs from SP to CUBE
Apply bind to all dial-peers when
CUBE has multiple interfaces.
Gig0/1 faces SP.
Translation rule/profile to strip the
access code (9) before delivering
the call to the SP

LAN Dial-Peer Configuration
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
codec g711ulaw
dtmf-relay rtp-nte
no vad
Outbound Dial-Peer for call legs from CUBE to CUCM
description *** Outbound LAN side dial-peer ***
destination-pattern 408527….$
session target ipv4:<CUCM_IP_Address>
codec g711ulaw
dtmf-relay rtp-nte
no vad
CUCM sending 9 (access code) + All
digits dialed
SP will be sending 10 digits (NANP)
based on your DID that is being
delivered to CUCM
Inbound Dial-Peer for call legs from CUCM to CUBE
Apply bind to all dial-peers when
CUBE has multiple interfaces. Gig0/0
faces CUCM.
Default codec is G729 if none is
specified
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing as the
traditional way to accommodate multiple trunks BRKUCC-2006 24

Step 3: SIP Normalisation
SIP incompatibilities arise due to:
• A device rejecting an unknown header (value
or parameter) instead of ignoring it
• A device expecting an optional header
value/parameter or can be implemented in
multiple ways
• A device sending a value/parameter that must
be changed or suppressed (“normalised”)
before it leaves/enters the enterprise to comply
with policies
• Variations in the SIP standards of how to
achieve certain functions
• With CUBE 10.0.1 SIP Profiles
can be applied to inbound SIP
messages as well
25BRKUCC-2006
More information at http://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/118825-technote-sip-00.html
Incoming Outgoing
INVITE
sip:5551000@sip.com:5060
user=phone SIP/2.0
INVITE
sip:5551000@sip.com:5060
SIP/2.0
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"
Add user=phone for INVITEs
Modify a “sip:” URI to a “tel:” URI in INVITEs
Incoming Outgoing
INVITE
tel:2222000020
SIP/2.0
INVITE
sip:2222000020@9.13.24.6:5060
SIP/2.0
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>"
CUBE
CUBE
SIP profiles is a mechanism to normalise or customise SIP at the
network border to provide interop between incompatible devices

Normalize Inbound SIP Message (Example 2)
request INVITE sip-header Diversion modify “sip:” sip:1234@
description Incoming/outgoing SP
voice-class sip profiles 700 inbound
Received:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
………
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 incoming
Dial-peer
Received:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
……….
Diversion: <sip:1234@9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
……….
m=audio 32278 RTP/AVP 18 8 101
a=rtpmap:0 PCMU/8000
CUBE
Requirement SIP Diversion header must include a user portion
SIP INVITE received by CUBE SIP INVITE CUBE expects
Enable Inbound SIP
Profile feature
voice service voip
sip
sip-profiles inbound
For Your
Reference

SIP Profile Tagging Enhancement
New rule tagging mechanism is being introduced
1. Insertion :
§ New rules can be inserted at any position i.e at the beginning, at the end or in between existing rules
by specifying rule tag number.
2. Deletion :
§ Rules can be deleted by giving no form of the rule with just the tag number.
3. Modification :
§ Any of the existing rules can be modified by specifying the rule tag number.
4. Duplication :
§ When a rule with an existing tag number is applied again, the rule will be over-written, without creating
any duplicate rules.
BRKUCC-2006 28

• For tagging the rules, an additional option of “rule <tag>” has been provided
SIP Profile Tagging – Configuration
CUBE(config)#voice class sip-profiles 1
CUBE(config-class)#?
VOICECLASS configuration commands:
exit Exit from voice class configuration mode
help Description of the interactive help system
no Negate a command or set its defaults
request sip request
response sip response
rule Specify the rule
CUBE(config-class)#rule ?
<1-1073741823> Specify the rule tag
before The rule to be inserted before
CUBE(config-class)#rule 1 ?
request sip request
The new
keyword “rule”
“tag” to be
provided with
rule keyword
BRKUCC-2006 29

• For inserting a rule between two rules, “before” option has been provided
SIP Profile Tagging – Configuration Cont’d
BRKUCC-2006 30
CUBE(config-class)#rule before ?
<1-1073741823> Specify the rule tag
CUBE(config-class)#rule before 3 ?
request sip request
• If rule <tag> option is used to configure a SIP Profile rule, then this rule can
be deleted by specifying just the tag number instead of specifying the entire
rule configuration.
CUBE(config-class)#no rule before <tag>
For inserting a rule
between two rules, the
new before keyword
is being introduced

Configuration Example
• For tagging the rules:
rule 1 request INVITE sip-header Contact Modify “(.*)” “1;temp=xyz”
rule 2 request INVITE sip-header Supported Add “Supported: ”
• For inserting a rule between two rules using “before” option:
rule before 2 request INVITE sip-header To Modify “(.*)” “1;temp=abc”
rule 2 request INVITE sip-header To Modify “(.*)” “1;temp=abc”
BRKUCC-2006 31
before
option
The new rule has
been inserted
between #1 and
#3

• Auto-Upgrade : Exec command - “voice sip sip-profiles upgrade”
• Suppose we have the following rules configured:
request INVITE sip-header Contact Modify “(.*)” “1;temp=xyz”
request INVITE sip-header Supported Add “Supported: ”
request REGISTER sip-header Contact Modify “(.*)” “1;temp=abc”
• After auto upgrade, the rules will be automatically upgraded as follows:
rule 3 request REGISTER sip-header Contact Modify “(.*)” “1;temp=abc”
BRKUCC-2006 32
Configuration Example Continued….

• Auto-Downgrade : Exec command - “voice sip sip-profiles downgrade”
• Suppose we have the following rules configured:
rule 3 request REGISTER sip-header Contact Modify “(.*)” “1;temp=abc”
• After auto downgrade, the rules will be automatically downgraded as follows:
request INVITE sip-header Contact Modify “(.*)” “1;temp=xyz”
request INVITE sip-header Supported Add “Supported: ”
request REGISTER sip-header Contact Modify “(.*)” “1;temp=abc”
BRKUCC-2006 33
Configuration Example Continued….

SIP Profile Support for Non-
Standard Headers

SIP Profile Support for Non-Standard Headers
§ Introducing support for adding/copying/removing/modifying non-
standard SIP headers using SIP profiles
§ A new 'WORD' option has been added to the SIP Profiles CLI chain to
allow the user to configure any non-standard SIP Header
CUBE(config-class)#request INVITE sip-header ?
Accept-Contact SIP header Accept-Contact
…….
Via SIP header Via
WORD Any other SIP header name
WWW-Authenticate SIP header WWW-Authenticate
CUBE(config-class)#request INVITE sip-header WORD ?
ADD addition of the header
COPY Copy a header
MODIFY Modification of a header
REMOVE Removal of a header
CUBE(config-class)#request INVITE sip-header WORD ADD “MyCustomHeader : Hussain Ali”
The new “WORD”
option for specifying
unsupported headers
BRKUCC-2006 35

• Advanced features on CUBE (Call Recording, Multi-Tenancy,
HA)
• CUBE Serviceability
Agenda

Understanding Dial-Peer Matching Techniques:
LAN & WAN Dial-Peers
• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending
and receiving calls to & from the PBX
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for
sending & receiving calls to & from the provider
38BRKUCC-2006
CUCM SIP Trunk ITSP SIP Trunk
CUBE
A
Outbound Calls Outbound WAN Dial-PeerInbound LAN Dial-Peer
IP PSTN
Inbound WAN Dial-PeerOutbound LAN Dial-Peer
Inbound Calls

Understanding Inbound Dial-Peer Matching Techniques
39BRKUCC-2006
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer
Inbound Calls
Outbound Calls
Match based on
Called Number
Match based on
Calling number
1
Match Based on URI
of an incoming
INVITE message
Default Dial-Peer = 0
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
2
3
4
Priority

Understanding Outbound Dial-Peer Matching Techniques
Match based on
Called Number &
carrier-id target
Match Based on URI
of incoming INVITE
message & carrier-id
target
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Content-Length: 226
........
Match based on URI
of an incoming
INVITE message
Match based on
Called number
CUBE
A
Outbound LAN Dial-Peer
IP
PSTN
Outbound WAN Dial-Peer
Inbound Calls
Outbound Calls
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
1
2
3
4
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
Priority
CSCua14749 – Carrier-id CLI not working on XE based
platforms
BRKUCC-2006 40

Destination Server Group
• Supports multiple destinations (session targets) be defined in a group and applied to
a single outbound dial-peer
• Once an outbound dial-peer is selected to route an outgoing call, multiple
destinations within a server group will be sorted in either round robin or preference
[default] order
• This reduces the need to configure multiple dial-peers with the same capabilities but
different destinations. E.g. Multiple subscribers in a cluster
41BRKUCC-2006
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
description Outbound DP
codec g711ulaw
dtmf-relay rtp-nte
session server-group 1

© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public 42BRKUCC-2006
Multiple Incoming Patterns Under Same
Incoming/Outgoing Dial-peer
SIP Trunk SP SIP Trunk
CUBE
IP PSTNA
(408)100-1010
(510)100-1010
(919)200-2010
(919)200-2000
(510)100-1000
(408)100-1000
voice class e164-pattern-map 300
e164 919200200.
e164 510100100.
e164 408100100.
description Inbound DP via Calling
incoming calling e164-pattern-map 300
codec g729r8
voice class e164-pattern-map 400
url flash:e164-pattern-map.cfg
description Outbound DP via Called
destination e164-pattern-map 400
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
Up to 5000 entries in a text file

Destination Dial-Peer Group
• Allows grouping of outbound dial-peers based on an incoming dial-peer, reducing
existing outbound dial-peer provisioning requirements
• Eliminates the need to configure extra outbound dial-peers that are sometimes
needed as workarounds to achieve desired call routing outcome
• Multiple outbound dial-peers are saved under a new “voice class dpg <tag>”. The
new “destination dpg <tag>” command line of an inbound voip dial-peer can be
used to reference the new dpg (dial-peer group)
• Once an incoming voip call is handled by an inbound voip dial-peer with an
active dpg, dial-peers of a dpg will then be used as outbound dial-peers for an
incoming call
• The order of outgoing call setups will be the sorted list of dial-peers from a dpg,
i.e, the destination-patterns of the outgoing dial-peers is not relevant for selection
BRKUCC-2006 43

Destination Dial-peer Group Configuration
voice class dpg 10000
description Voice Class DPG for SJ
dial-peer 1001 preference 1
dial-peer 1003
!
description Inbound DP
incoming called-number 1341
destination dpg 10000
BRKUCC-2006 44
!
!
session target ipv4:10.1.1.31. Incoming Dial-peer is first
matched 2. Now the DPG associated with
the INBOUND DP is selected

External/PSTN Call Recording Options
• CUBE Controlled (Dial-peer based ORA)
• Based on Open Recording Architecture, metadata sent in Cisco Proprietary format from CUBE to Recorder
• Dial-peer controlled, IP-PBX independent
• Source of recorded media (RTP only) is always CUBE (External calls only). For SRTP-RTP calls, apply
media forking CLI on the RTP leg only.
• Records both audio and video calls and supported with CUBE HA (Inbox or box-2-box)
• CUBE Controlled (Dial-peer based SIPREC)
• Based on SIPREC (RFC 6341, 7245, Metadata-draft-17, Protocol-draft-15), CUBE sends metadata in XML
format
• Dial-peer controlled, IP-PBX independent
• Source of recorded media (RTP only) is always CUBE (External calls only). For SRTP-RTP calls, apply
media forking CLI on the RTP leg only.
• Records both audio and video calls and supported with CUBE HA (Inbox or box-2-box)
• CUCM NBR (Network Based Recording)
• CUCM Controlled, requires CUCM 10+ and UC Services API be enabled on CUBE
• Recording triggered by CUCM and this mode records only Audio calls
• Source of Recorded Media can be CUBE or Endpoint (BiB), CUBE as source desired for PSTN calls
46BRKUCC-2006

CUBE Controlled Recording Option – Media Forking
• Call agent
independent
• Configured on a per
Dial-peer level to fork
RTP
Cisco MediaSense
(authentication disabled w/o UCM)
Cisco Search/Play demo app or Partner
Application
media class 9
recorder parameter
media-recording 950
description dial-peer that needs to be forked
media-class 9
description dial-peer pointing to MediaSense
destination-pattern 9999 ! Dummy
session transport tcp
session target ipv4:<Mediasense_IP>
! Bind on this DP mandatory
• 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
• With XE 3.10.1, Video calls supported
and CUBE HA for audio calls
SIPSIP
SIP
A
SP SIP
CUBE
RTP RTP
MediaSense
Needs to
match
Dial-peer based – Open Recording Architecture (ORA)
Cisco Proprietary Metadata
BRKUCC-2006 47

CUBE Controlled Recording Option - SIPREC
• SIP Profiles can
additionally be used to
forward 3rd party IP PBX
Call Identifier to the
Recorder for Correlation
SIPREC Compliant Recorder
• SIP is used as a protocol between CUBE and the recording server, where CUBE acts as the recording
client and any third party recorder acts as the recording server
• Along with SDP, metadata information is passed by CUBE to the recording server in XML format
SIPSIPA
SP SIP
CUBE
RTP RTP
Recorder
Dial-peer based – SIPREC Standard
XML Metadata
• Metadata includes the
communication session details of
audio or video calls and also
identifies the participants of the call
media class 9
recorder parameter siprec
media-recording 950
description dial-peer that needs to be forked
media-class 9
description dial-peer pointing to MediaSense
destination-pattern 9999 ! Dummy
session transport tcp
session target ipv4:<Mediasense_IP>
! Bind on this DP mandatory
Needs to
match
BRKUCC-2006 48

CUCM (10.X or later) Controlled Recording
49BRKUCC-2006
1. Enable HTTP on IOS
ip http server
http client persistent
2. Enable the API on IOS
uc wsapi
source-address [IP_Address_of_CUBE]
3. Enable XMF service within the API
provider xmf
remote-url 1 http://CUCM:8090/ucm_xmf
no shutdown
Gateway/CUBE Recording
Enabled
1. 2.
3.
4.
5.
[1] – [3]: An external call is answered by user with IP phone
[4] – [5]: CUCM sends forking request over HTTP to CUBE, which
sends two media streams towards the Recording Server
UC Services API – Network Based Recording
• Recording not preserved on failover in CUBE HA
• Selective Recording, Mobile/SNR/MVA Calls
• Recording Call Preservation
Now Supports Inbound CVP (Survivability.tcl) Call Recording
[IOS 15.6(1)T, IOS-XE 3.17]

Multiple Non-Authenticated SIP
Trunks on a CUBE

Non-Authenticated SIP Trunking to more than one
Service Provider
A
TDM PBX
SRST
CME
MPLS
Enterprise
Branch Offices
Enterprise
Campus
Active
CUBE
SIP SP-1
(10.10.10.2)
SIP SP-2
(20.20.20.2)
Large enterprises are deploying more
than one SIP Trunk provider for:
• Alternate call routing
• Load balancing
description “Secondary path to SIP SP-2”
destination-pattern 91[2-9]..[2-9]......
preference 2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback2
voice-class sip bind media source-interface loopback2
interface loopback1
ip address 10.10.10.1 255.255.255.0
interface loopback2
ip address 20.20.20.1 255.255.255.0
description “Primary path to SIP SP-1”
destination-pattern 91[2-9]..[2-9]......
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback1
voice-class sip bind media source-interface loopback1
SIP SP-1’s
network
SIP SP-2’s
network
NOTE: Dual SPs can be used for outbound calls, but to
be utilised for inbound calls, arrangements between
SPs required

Multiple Authenticated/Registered
SIP Trunks on a CUBE

Multiple Instances of SIP-UA on a CUBE
BRKUCC-2006 53
Existing Implementation, prior to IOS 15.6(2)T and IOS-XE 16.3.1
• CUBE Configuration generally consists of
• Global – Everything under voice service voip
• Call Routing – Dial-peers (Any configuration under dial-peers always overrides Global config)
• SIP User Agent Config – Everything under sip-ua, applicable globally on the platform
• No provision to configure specific bind/credentials/outbound proxy for
different registrar
• No provision to configure specific configs (e.g. timers, retry) for
different tenants
• Unable to handle authentication challenge for more than one trunk that
have the same SIP realm

Introducing Tenants on CUBE
BRKUCC-2006 54
• Every Registrar/User Agent/ITSP connected to CUBE can be
considered a Tenant to CUBE
• Allows specific global configurations (CLI under sip-ua) for multiple
tenants such as specific SIP Bind for REGISTER messages
• Allows differentiated services for different tenants

“Voice class Tenant” Overview
Prior to Multi Tenancy
sip-ua
registrar 1 ipv4:60.60.60.60:9051 expires 3600
credentials username aaaa password 7 06070E204D realm aaaa.com
credentials username bbbb password 7 110B1B0715 realm bbbb.com
voice service voip
outbound-proxy ipv4:10.64.86.35:9057
bind control source-interface GigabitEthernet0/1
BRKUCC-2006 55
With Voice Class Tenant (Multi-Tenancy)
voice class tenant 1
credentials username bbbb password 7 110B1B0715 realm bbbb.com
E164 - aaaa
E164 - bbbb
Registrar - 1
Registrar - 2
E164 - aaaa
E164 - bbbb
Registrar - 1
Registrar - 1
OB Proxy 1 & Bind-1
OB Proxy 2 & Bind-2
• Most configs under “sip-ua” and “voice service voip” added in “voice class tenant <tag>”,
e.g. Registrar and Credentials CLI under tenant using different bind and outbound proxy
Global OB Proxy and Bind

Authenticating Multiple trunks with same Realm
BRKUCC-2006 56
With Voice Class Tenant (Multi-Tenancy)
registrar 1 dns:cisco.com expires 3600
credentials number +1234 username aaaa@cisco password 0 AAAA realm cisco.com
authentication username aaaa@cisco password 7 AAAA realm cisco.com
registrar 1 dns:cisco.com expires 3600
credentials number +6789 username bbbb@cisco password 0 BBBB realm cisco.com
authentication username bbbb@cisco password 7 BBBB realm cisco.com
• Requirement : To register two different authenticated numbers/usernames to different
registrars, but with the same realm
• Prior to IOS 15.6(2)T / IOS-XE 16.3.1, CUBE could register multiple trunks only with
different realms as the “authentication” command only accepted different realms. If the
realms were the same, it just overwrote the username and password
• Now each credential/authentication pair can be defined under its own voice class tenant
so that the same realm can be used for authentication

Configuring Voice Class Tenant
BRKUCC-2006 57
• Configure voice class tenant
credentials number bbbb username bbbb password 7 110B1B0715 realm bbbb.com
bind media source-interface GigabitEthernet0/0
copy-list 1
early-offer forced
• Apply tenant to the desired dial-peer
session target ipv4:10.64.86.35:9051
session transport udp
voice-class sip tenant 1 Apply Tenant to a Dial-peer
Add new voice class tenant

Introduction to Multi-VRF
BRKUCC-2006 59
ATT
VZN
SPT
• Virtual Routing and Forwarding (VRF) is an IP technology that allows for multiple
instances of a routing table to coexist on the same router at the same time as
opposed to a single global route table, allowing for multiple virtual networks within a
single network entity to isolate between media and data virtual networks
• Multi-VRF allows for the use of only one router to accomplish the tasks that
multiple routers usually perform
• Prior to IOS 15.6(2)T / IOS-XE 16.3.1, CUBE only supports a single VRF for Voice
[voice vrf vrfname]

Multi-VRF and CUBE Enterprise
BRKUCC-2006 60

Multi-VRF Aware Call Routing on CUBE
• CUBE allows intra and inter VRF routing of voice and video calls without the need of Route
Leaks improving security at the network level
• Overlapped IP addressing and Dial Plan with Multi VRF feature provides seamless integration
of networks
• Dial-peer bind statements are mandatory as the VRF association to a dial-peer is based upon
the interface sip bind and both Control and Media on a dial-peer has to bind with the same
VRF. Default incoming dial-peer (dial-peer 0) match is not supported with VRF
BRKUCC-2006 61

Understanding Inbound Dial-Peer Matching Techniques
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer
Inbound Calls
Outbound Calls
Match based on
Called Number
Match based on
Calling number
1
Match Based on URI
of an incoming
INVITE message
Default Dial-Peer = 0
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Content-Length: 226
........
2
3
4
Priority
0
Filter dial-peers based
on incoming VRF if
configured and then 1
to 3 below
BRKCOL-2125 62

Multi-VRF Feature Restrictions
• CUBE + CME co-located with VRF and TDM-SIP gateway are not supported
• IPV6 with VRF is not supported on CUBE. Only IPv4 is supported with VRF
• Multi-VRF calls across CUBE are supported in SIP-SIP flow-through mode only
and not supported in flow-around mode.
• Media Anti-trombone is not supported with VRF
• Legacy global voice vrf and Multi VRF doesn’t co-exist. Customers using global
voice vrf have to remove the CLI in order to use Multi VRF feature
• UC Services API (CUCM NBR Recording) is not VRF aware. Works globally for all
call recordings and will not separate the call notification on a per VRF basis
• Dial-peer group feature or COR (Class of Restriction) lists can be used to restrict
call routing to the same or group of VRFs (e.g. Overlapping Dial plans)
• The DSP resources are a global pool and not reserved on a per VRF basis. It is
used on a first come first serve basis
BRKUCC-2006 63
For Your
Reference

CUBE Multi VRF - Basic Configuration
CUBE
VRF 1 VRF 2
ip vrf vrf1
rd 1:1
interface GigabitEthernet0/0/0
ip address 7.44.44.13 255.255.0.0
ip vrf forwarding vrf1
voice-class sip bind all interface GigE0/0/0
ip vrf vrf2
rd 2:2
ip address 6.44.44.13 255.255.0.0
1. Configure VRF
2. Apply VRF under the interface/sub-interface
3. Bind the VRF associated interface to the dial-peer (VRF association by dial-peer bind CLI)
• Up to 54 different VRFs supported in 15.6(3)M and IOS-XE 16.3.1 or later releases
Gig0/0/0 Gig0/0/1
BRKUCC-2006 64

CUBE Multi VRF – Inbound dial-peer match
ip vrf vrf1
rd 1:1
!
ip address 7.44.44.13 255.255.0.0
!
ip vrf vrf2
rd 2:2
!
ip address 6.44.44.13 255.255.0.0
!
§ Inbound match based on VRF where SIP INVITE received
§ For VRF 1, dial-peer 100 is matched
§ For VRF 2, dial-peer 200 is matched
CUBE
VRF 1 VRF 2
INVITE
sip:2000@7.44.44.13
INVITE
sip:2000@6.44.44.13
BRKUCC-2006 65

CUBE Multi VRF – Inter/Intra VRF Routing
ip vrf vrf1
rd 1:1
ip address 7.44.44.13 255.255.0.0
session-target ipv4: 10.1.1.1
ip vrf vrf2
rd 2:2
ip address 6.44.44.13 255.255.0.0
session-target ipv4:10.2.2.2
incoming called-number 3000Intra VRF
Routing
Inter VRF
Routing
CUBE
VRF 1 VRF 2
VRF1
INVITE sip:3000@7.44.44.13 INVITE sip:3000@6.44.44.13
VRF2VRF2

CUBE Multi VRF – Routing w/ Overlapped Dial Plan
Route Inter or
Intra VRF calls
based on
outbound
dial-peer groups
ip vrf vrf1
ip address 7.44.44.13 255.255.0.0
voice class dpg 100
destination dpg 100
ip vrf vrf2
ip address 6.44.44.13 255.255.0.0
voice class dpg 200
destination dpg 100
INVITE
sip:2000@7.44.44.13
INVITE
sip:2000@6.44.44.13
INVITE
sip:2000@10.1.1.1
INVITE
sip:2000@10.1.1.1
VRF1 VRF2
VRF1

CUBE Multi VRF – Call Routing w/ Overlapped IP
ip vrf vrf1
ip address 7.44.44.13 255.255.0.0
destination dpg 100
voice class dpg 100
ip vrf vrf2
ip address 7.44.44.13 255.255.0.0
destination dpg 200
voice class dpg 200
INVITE
sip:1000@7.44.44.13
INVITE
sip:2000@7.44.44.13
INVITE
sip:1000@10.1.1.1
INVITE
sip:2000@10.1.1.1
VRF1 VRF2
Overlapped
local IP

CUBE HA Design Considerations on ISR-G2 for Box-to-Box
Redundancy
70BRKUCC-2006
• Anytime a platform is reloaded in a CUBE-HA relationship, it always boots up as Standby
• All signaling/media is sourced from/to the Virtual IP Address
• Multiple Traffic (SIP/RTP) interfaces (Gig0/0, Gig0/1) require preemption and interface tracking
• HSRP Group number should be unique to a pair/interface combination on the same L2
• All interfaces of the same group have to be configured with the same priority
• No media-flow around or UC Services API (CUCM NBR) support for CUBE HA

• Lower IP Address for ALL the interfaces (Gig0/0, Gig0/1, Gig0/2) should be on the same platform,
which is used as a tie breaker for the HSRP Active state
• Multiple HSRP Groups/Interfaces/sub-interfaces can be used on either LAN or WAN side
• Upto 6 multimedia lines in the SDP are checkpointed for CUBE HA
• SDP Passthru (upto 2 m-lines) calls are also checkpointed starting IOS 15.6(1)T
• TDM or SRST or VXML GW cannot be collocated with CUBE HA
• Both platforms must be connected via a physical Switch across all likewise interfaces for CUBE HA to
work, i.e. Gig0/0 of CUBE-1 and CUBE-2 must terminate on the same switch and so on
Redundancy – Cont’d
71BRKUCC-2006

• Cannot have WAN terminated on CUBEs directly or Data HSRP on either side. Both Active/Standby
must be in the same Data Centre
• Both the CUBEs must be running on the same type of platform and IOS version and identical
configuration. Loopback interfaces cannot be used for bind as they are always up. Sub-interfaces are
supported for all interfaces. Port Channels are supported for all interfaces from IOS 15.6(3)M
• CUBE HA only checkpoints SIP/RTP Traffic. Support for Survivability.tcl preservation was added in
15.6(2)T for CVP deployments
72BRKUCC-2006

• LTI based transcoding called flows including SRTP/RTP interworking preserved starting 15.5(2)T.
Requires same PVDM3 chip capacity on both active and standby in the same slot/subslot. CPA calls
(prior to being transferred to the agent), SCCP based media resources, Noise Reduction, ASP,
transrating calls are not checkpointed
• SRTP - RTP, SRTP - SRTP and SRTP passthru supported on ISR-G2
73BRKUCC-2006

• CUBE HA with HSRP is supported with VRFs configured
• Traffic interfaces (SIP/RTP) can have VRFs configured but HSRP interface [ipc zone default config –
Gig0/2 above] cannot have any VRF associated with it. This means for every CUBE HA deployment
where VRFs are being utilized for SIP/RTP interfaces, at least three interfaces are required. Otherwise,
any of the LAN interfaces (Gig0/0 above) can be used as an HSRP interface
• VRF ID’s will be check pointed for the calls before and after switchover. VRF Configurations in both
active and standby routers have to be identical. This includes VRF based rtp port range as well
• Upon failover, the previously ACTIVE CUBE goes through a reload by design, preserving
signalling/media. Thus, running config should always be saved to avoid losing it due to the
reload
74BRKUCC-2006

CUBE Configuration on ISR-G2 Box-to-Box Redundancy
75BRKUCC-2006
CUBE-1> enable
CUBE-1# configure terminal
CUBE-1(config)# ip vrf LAN-VRF
CUBE-1(config)# rd 1:1
CUBE-1(config)# ip vrf WAN-VRF
CUBE-2> enable
CUBE-2# configure terminal
CUBE-2(config)# ip vrf LAN-VRF
CUBE-2(config)# ip vrf WAN-VRF
Configure VRFs
on the platform
(if applicable)
CUBE 1 CUBE 2

76BRKUCC-2006
interface GigabitEthernet0/0
description “Enterprise LAN”
ip vrf forwarding LAN-VRF
ip address 10.10.1.12 255.255.255.0
standby version 2
standby 1 ip 10.10.1.13
standby delay minimum 30 reload 60
standby 1 preempt
standby 1 track 2 decrement 10
standby 1 priority 50
Inside interfaces :
HSRP group 1
VRF ID : LAN-VRF
(if applicable)
Interface can be
utilised as an HSRP
interface if no VRFs
are required or
configured
CUBE 1 CUBE 2
description “Enterprise LAN”
ip vrf forwarding LAN-VRF
ip address 10.10.1.11 255.255.255.0
standby version 2
standby 1 ip 10.10.1.13
standby 1 preempt

77BRKUCC-2006
description “Enterprise WAN”
ip vrf forwarding WAN-VRF
ip address 128.107.66.77 255.255.255.0
standby version 2
standby 10 ip 128.107.66.79
standby 10 preempt
description “Enterprise WAN”
ip vrf forwarding WAN-VRF
ip address 128.107.66.78 255.255.255.0
standby version 2
standby 10 ip 128.107.66.79
standby 10 preempt
Outside
interfaces :
HSRP group 10
VRF ID : WAN-
VRF (if
applicable)
CUBE 1 CUBE 2

78BRKUCC-2006
description “HSRP Interface”
ip address 1.1.1.1 255.255.255.0
standby version 2
standby 100 ip 1.1.1.3
standby 100 preempt
standby 100 name CUBEHA
!
track 1 interface Gig0/0 line-protocol
description “HSRP Interface”
ip address 1.1.1.2 255.255.255.0
standby version 2
standby 100 ip 1.1.1.3
standby 100 preempt
standby 100 name CUBEHA
!
Configure Interface
Tracking (for line protocol
on corresponding
interfaces of the platform
HSRP interfaces :
HSRP group 100
CANNOT HAVE VRFs
associated
CUBE 2CUBE 1

79BRKUCC-2006
redundancy inter-device
scheme standby CUBEHA
voice service voip
mode border-element
redundancy
ipc zone default
association 1
no shutdown
protocol sctp
local-port 5000
local-ip 1.1.1.1
remote-port 5000
remote-ip 1.1.1.2
redundancy inter-device
scheme standby CUBEHA
voice service voip
mode border-element
redundancy
ipc zone default
association 1
no shutdown
protocol sctp
local-port 5000
local-ip 1.1.1.2
remote-port 5000
remote-ip 1.1.1.1
Define Redundancy scheme: Creates
interdependency b/w CUBE
redundancy & HSRP
Turn on CUBE Redundancy
HSRP Interface - IPC configuration :
Allows the ACTIVE CUBE to tell the
STANDBY about the state of the calls.
CONFIG SHOULD BE APPLIED on the
LAN SIDE (to avoid SPLIT BRAIN) and a
NON-VRF associated interface
CANNOT HAVE VRFs
associated with this interface
CUBE 1 CUBE 2

80BRKUCC-2006
description TO SERVICE PROVIDER
destination-pattern 9T
session target ipv4:y.y.y.y
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
!
description TO CUCM
destination-pattern 555….
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
!
ip rtcp report interval 3000
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 86400
Bind traffic destined to the outside (SP SIP trunk)
to the outside Physical interface.
This ensures that all RTP and SIP packets are
created with the virtual IP associated with the
respective physical interface.
CUBE HA does not work with loopback interfaces
as they are always up
Configuration on Active and Standby
Bind traffic destined to the inside (CUCM or IP
PBX) to the inside Physical interface.
This ensures that all RTP and SIP packets are
created with the virtual IP associated with the
respective physical interface.
Configure media inactivity feature to clean up any
calls that may not disconnect after a failover

CUBE HA Design Considerations on ASR1K/ISR-4K/vCUBE
for Box-to-Box Redundancy
81BRKUCC-2006
• Uses Redundancy Group (RG) Infrastructure Protocol
• GE0/0/0 and GE0/0/1 are referred to as traffic (SIP/RTP) interfaces and GE0/0/2 is RG (Redundancy
Group) Control/data interface
• Starting IOS-XE 16.3.1, Port channel is supported for both RG Control/data and traffic interfaces
• When configuration is applied and saved, the platform must go through a reload cycle
• RG Control/Data Interfaces (GE0/0/2) must be connected through a Switch and not a Crossover Cable

Additional Supported options for CUBE HA
CUBE-1
CUBE-2
CUCM
WAN
Edge
PortChannel34
Gig0/0/3
PortChannel34
Gig0/0/4
Gig0/0/3
Switch A Switch B
CUBE
CUBE
Gig0/0/2.100 – ITSP 1
Gig0/0/2.100
Gig0/0/0
Gig0/0/0
Gig0/0/4
redundancy
rii 3
redundancy
rii 2
Switch C
Gig0/0/2.200
Gig0/0/2.200 – ITSP 2
Gig0/0/1
Gig0/0/1
PortChannel2PortChannel2
Switch D Switch E
redundancy
rii 1
ITSP 1
ITSP 2
vPC
vPC
82
• The RG control data interfaces can be a sub interface that is part of the same port channel used for voice traffic. This will go to switch D and E
thereby eliminating the need for additional switches for RG control/data. This is provided there is sufficient bandwidth for voice + RG
data/control on the port channel (for example when using 10G)
• Multiple ITSPs or multiple trunks from the same ITSP can be terminated on the same CUBE ENT HA (ISR G2, ISR 4K, ASR 1K, vCUBE) pair
• Port Channel(s) can be used on the WAN/ITSP side as well as shown for the LAN side in the above diagram with L2 and CE router redundancy

Video Suppression
• When CUBE receives video capabilities as part of SDP, it passes them across by
default
• This feature adds a mechanism on CUBE to allow only audio and image (for T.38
fax) media capabilities and drop all other media capabilities like video,
application m-lines etc. while routing calls to service providers
»Only supported for SIP-SIP calls not in SDP Passthru mode
BRKUCC-2006 84
A SP
SIP
INVITE w/ audio,
video, application
CUBE(config)#voice service voip
CUBE(conf-voi-serv)#sip
CUBE(conf-serv-sip)#audio forced
CUBE(conf-serv-sip)#dial-peer voice 100 voip
CUBE(conf-serv-sip)#description “Outgoing Dial-peer”
CUBE(config-dial-peer)#voice-class sip audio forced
SBCCUBE
INVITE w/
audio only
Video
Endpoints

Cisco Unified Border Element Basic Call Flow
1. Incoming VoIP setup message from originating endpoint
2. This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD,
DTMF method, protocol, etc.
3. Match the called number to outbound VoIP dial peer 2
4. Outgoing VoIP setup message
87BRKUCC-2006
Incoming VoIP Call Outgoing VoIP Call
codec g711ulaw
codec g711ulaw
Originating
Endpoint -
1000
Terminating
Endpoint –
2000
CUBE
voice service voip
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323

CUBE
1000
1.1.1.1
2000
2.2.2.2
Incoming VoIP Call Leg
Matches an Incoming Dial-peer
Outgoing VoIP Call Leg
Matches an Outbound Dial-peer
INVITE /w SDP INVITE /w SDP
100 TRYING100 TRYING
VRF1 – 10.10.10.10 20.20.20.20 – VRF2
c= 1.1.1.1
m=audio abc RTP/AVP 0
c= 20.20.20.20
m=audio xxx RTP/AVP 0
180 RINGING180 RINGING
200 OK
200 OK c= 2.2.2.2
m=audio uvw RTP/AVP 0c= 10.10.10.10
m=audio xyz RTP/AVP 0
RTP (Audio)
ACK
ACK
1.1.1.1 10.10.10.10 20.20.20.20 2.2.2.2
Understanding the Call Flow
BYE BYE
200 OK
200 OK
88BRKUCC-2006

Basic Show Commands for Active Calls
89BRKUCC-2006
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections

90BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

91BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

92BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

93BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

94BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

95BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

96BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

97BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

98BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

99BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

100BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

101BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

102BRKUCC-2006
dur 00:00:14 tx:0/0 rx:0/0
dur 00:00:14 tx:0/0 rx:0/0
SIP call-legs: 2
H323 call-legs: 0
SCCP call-legs: 0
Total call-legs: 2
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1
2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2

Total Number of Active Concurrent Calls
Router# show call active total-calls
Total Number of Active Calls : 10
Total Number of Active Calls
§ A single call can have multiple call-
legs. To determine the total number
of active calls from call-legs is
challenging
§ CLI added to display the value of
current number of active
(connected) calls on CUBE
§ The table defines the relation
between call-legs and number of
active calls
Call Flow Call-legs
Connected
call
Basic call (audio/video) 2 1
Transferred call (Refer
handling)
3 2
Transcoded call (SCCP) 4 1
Calls after rotary/hunt 2 + x 1
Forwarded calls (CUBE
handling)
3 1
Forked call (media forking) 3 2
Forked call (signalling forking) 2 1

Avoiding Non-Call-Context Debug Logs
• Many times SIP debugs contain unrelated debugs that are not useful in
debugging issues related to call failures
• Starting CUBE 10.0.1, non-call-context debugs will not be printed when
debug ccsip is issued
• This applies to messages originating from CUBE. Non-call context
INBOUND messages towards CUBE will still be printed when
debug ccsip is issued.
• If a message is not part of any call, that debug will not be printed
• Affected messages: OPTIONS, REGISTER, SUBSCRIBE/NOTIFY
• To see the above OUTBOUND messages in debugs, issue the following
command
debug ccsip non-call

Debugging Made Easier
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 >
Categorise Debugs based on Functionality
§ Categorisation 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

Debugging Made Easier
CUBE# show cube debug category codes
Categorise 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
|-----------------------------------------------

• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Management & Troubleshooting
Agenda

Introducing Cisco Business
Edition 4000 (BE4K)

BE4K – Simple, cloud-managed office communications
BRKUCC-2006 109
Features
ü Packaged Collaboration Offer - Voice and Voicemail
solution for SMBs (200 users)
ü Based on ISR4321 router running CUCME and virtualized
CUE.
ü Support for the latest IP Phones (78xx, 88xx, Jabber,
ATA-190)
ü Rapid Installation – Preconfigured before shipment
ü Smart licensing
ISR 4321

BE4K – Contd.
BRKUCC-2006 110
Voice & Voicemail
ü Cisco OEM Cloud Management.
ü Cloud Deployed - configuration, administration and
management.
ü Rich Feature Set – Auto-attendant, Flexible call
routing, Park, Hold, Transfer, Hunt groups,
Conferencing, Voicemail.
ü Single or multiple sites.
ü Fixed configuration SKU for ease of ordering.
ü Migration options include BE6000 / BE7000, Cisco
Spark. No throw-away capex.
ü Add-on Spark and WebEx for messaging and
meetings.

Key Takeaways
• Enterprise SBC (Cisco Unified Border Element - CUBE) is an essential
component of a UC solution:
• Over 18,000 Enterprise customers all over the Globe
• Proven interoperability with 3rd party PBX vendors and different service providers
around the world (more than 165 countries)
• Complete feature Presentations, Lab Guide, & Application Notes :
»https://cisco.box.com/cube
• Day long, hands-on lab sessions next week free of charge at Cisco Offices
Title: Deploying Enterprise SIP Trunks with CUBE (Cisco Unified Border
Element) and CUCM (Cisco Unified Communications Manager)
• March 13th – Cisco North Sydney – 177 Pacific Hwy
• March 15th – Cisco Melbourne – 101 Collins Street
• March 17th – Cisco Auckland – 152 Fanshawe Street
• Contact Hussain Ali you are interested in attending
111BRKUCC-2006

Continue Your Education
• Demos in the Cisco campus
• Walk-in Self-Paced Labs
• Lunch & Learn
• Meet the Engineer 1:1 meetings
• Related sessions
112BRKUCC-2006

Complete Your Online Session Evaluation
114BRKUCC-2006
Learn online with Cisco Live!
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Cisco Live! :: Deploying SIP Trunks with Cisco Unified Border Element (CUBE/vCUBE) Enterprise | 2017

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