Brkcol 2125

SIP Trunk Design with Cisco
Unified Border Element

(CUBE/vCUBE) Enterprise
and SRST Co-location
Hussain Ali, CCIE# 38068 (Voice, Collaboration)
Technical Marketing Engineer
Dilip Singh, CCIE# 16545 (Collaboration)
Technical Leader
BRKCOL-2125
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#CiscoLiveLA © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 2
Agenda
• CUBE Overview and SIP Trunk Sizing
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE + SRST Colocation
• Futures & Key Takeaways
#CiscoLiveLA BRKCOL-2125 © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 3
CUBE Overview and Sizing
Collaboration Deployment
DEMARC
Enterprise LAN ITSP WAN (SIP Provider)
PSTN (PRI/FXO)
Unified CM
TDM Backup
(Not available in
10.10.1.20
10.10.1.21 vCUBE)
66.77.37.2
Gig0/0
PSTN
Gig0/1
CUBE 128.107.214.195
SIP
DEMARC
H.323
RTP
CUBE (Enterprise) Product Portfolio
50-150 ASR 1004/6 RP2
Introducing CUBE on ASR 1002-X
50-100 CSR
ASR 1001-X
vCUBE [Performance
dependent on vCPU and ISR 4451-X
memory] ISR 4431
20-35 3900E Series ISR-G2
(3925E, 3945E)
CPS
ISR 4351
17
3900 Series ISR-G2 (3925, 3945)

ISR-4K (4321, 4331)
8-12
2900 Series
ISR-G2 (2901, 2911, 2921, 2951) Note: SM-X-PVDM module
<5
supported on XE3.16 or later for
ISR 4K platforms
800 ISR
4 <50 500-600 900-1000 2000-2500 4000 4500-6000 7000-10,000 12K-14K 14-16K
Active Concurrent Voice Calls Capacity

CUBE Software Release Mapping
ISR G2 ASR 1K / ISR-4K/vCUBE (CSR)
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
EOL EOL EOL 12.0.0 16.6.1 July 2017
EOL EOL EOL 12.0.0 16.7.1 Nov 2017
EOL EOL EOL 12.1.0 16.8.1 Mar 2018
EOL EOL EOL 12.2.0 16.9.1 July 2018
EOL EOL EOL 12.5.0 16.10.1 Nov 2018
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
CUBE Interoperability
Cisco Interoperability Portal: • Validated with Service
www.cisco.com/go/interoperability Providers World-Wide
• Independently Tested
with 3-Party PBXs in
tekVizion Labs
• Standards based
Verified by
Proven Interoperability and Interworking with

Service Providers Worldwide
Sizing Enterprise CUBE
deployments
NOTE : Sizing information is only intended as a
guideline. Actual session count will vary based
on the number of features turned on the
ISR/ASR/CSR along with CUBE and the IOS-XE
version being used.
CUBE Session Capacity Summary For Your
Reference
Platform CUBE SIP-SIP IPT Sessions (Audio)

NanoCUBE (8XX and SPIAD Platforms) 15 - 120
2901 – 4321 100
2911 – 2921 200 – 400
4331 500 • Flow thru
2951 600 • RTP-RTP
3925 – 3945 800 – 950
• IPT
4351 1000
3925E – 3945E 2100 – 2500 • 711-711
4431 3000
4451 6000
ASR1001-X 12000
ASR1002-X 14000
ASR1004/1006 RP2 16000
CUBE ENT on ISR 4K Series
Audio Session Capacity with Additional Features: XE16.6 and later
CPS1 Total # of SW CUBE + CUBE + Xcoding2

Flow-thru MTP3 SW
Platform (Memory4)
calls5 MTP3 Xcoded Calls w/ Simultaneous Non-
(Estimates) PVDM4-256 Xcoded Calls
4 100 250 71 100 CUBE FT sessions
4321 (4GB) maxed
4331 (4GB) 8 500 600 250 128 372
4351 (4 GB) 10 1000 1000 500 128 872
4431 (8 GB - CP) 15 3000 1500 750 128 2872
4451 (8 GB - CP) 40 6000 3000 1500 128 5872
1. CPS and session counts listed are independently tested. Session capacities [Total # of Flow-thru calls] can be achieved at about half
the CPS listed in the Calls Per Second (CPS) column.
2. Transcoding is limited by DSP capacity. ISR4K has one PVDM4 motherboard slot and 128 sessions is based on a single PVDM4-256.
(G729r8-G711)
3. S/W MTP numbers are for standalone SW MTP sessions and CUBE+SWMTP numbers are for the maximum number of CUBE sessions
supported with each session utilizing a SW MTP on the same platform
4. All tests were done with 4 GB of RAM with the exception of 4451/4431 where 8 GB was used [8GB CP and 2GB DP]
5. Total calls are derived with 180 seconds call hold time
CUBE Sizing Guidelines
• All deployments for CUBE Ent must be done with 16GB of memory for ASR1K series,
8 GB (Control Plane memory) for ISR4400 series, 4 GB for ISR4300 series, and 2 GB
for ISR G2 series
• Session count (end to end calls through CUBE) is dependent on the amount of
memory in the box. Numbers listed in the datasheet assume above memory
requirements are being satisfied
• CPS is dependent on the CPU of the platform
• Complex call flows (Cisco UCCE) can reduce CUBE CPS and session count by upto 75%
on ISR 4K/ASR1K series
• CUBE HA has less than 5% impact on number of sessions under full load
CUBE Sizing Guidelines
• SRTP with SIP TLS : Numbers will vary based on crypto algorithm and codec used
• SIP Header manipulation through SIP profiles has less than 5% impact on number of
sessions. Impact of SDP manipulation will be slightly higher compared to SIP headers.
For example, 6% for changing the codec order in the m-lines
• Media forking for call recording can have a 50% impact on IPT session count
regardless of the call type (IPT or UCCE) being recorded
• ASR1K Series : Recommendation is to use an RP2 based platform with a minimum of
ESP20 (1004/1006). Additionally 1001-X and 1002-X are recommended for CUBE
deployments instead of 1001 and 1002 respectively. ASR1001, ASR1002, ESP 5, ESP
10, RP1, and SIP10 will not be supported in the IOS XE 16 releases
CUBE Performance : DSP Considerations
• Transcoded (Xcoded) Session Count: is based on the DSP capacity for all ASR1K and ISR 4K series
• An ISR4K with a single motherboard PVDM slot can have a maximum of PVDM4-256. Each PVDM4-256
supports
1. 256 transcoding sessions if using a Low Complexity (LC) codec. E.g. g711alaw <-> g711ulaw
2. 192 xcoding sessions if using a Medium Complexity (MC) codec. E.g. g729abr8/g729ar8 <-> g711
3. 128 sessions if using High Complexity (HC) codec. E.g. g729r8/g729br8 <-> g711 (64 w/PVDM4-128)
• By default IOS-XE devices have codec complexity set to flex to have the max call density and as such they do
g729ar8 even when the CLI says codec g729r8 under a dial-peer or voice class codec
• On ISR 4K series you can have more DSP capacity by using an SM-X-PVDM module or by oversubscribing a
T1/E1 NIM with extra DSP channels by using a larger DSP PVDM4 than required for module density of voice
T1 ports. These extra DSP channels can be used for voice IP services (xcoding, conferencing, MTP) and can be
pooled together with the on-board PVDM4 and extra DSP channels on separate T1/E1 NIM slot to create a
larger DSPfarm
U BE Session is
Sample ISR4K CUBE Sizing TOTAL C Required in th
License
s
• An enterprise is looking for an SBC to support 800 IP setup: 9
00
telephony sessions and an additional 100 Cisco Contact
Center agent calls. All CC calls must be recorded. G711 is CPS CUBE
Platform
used throughout but 50 remote agent phones will require Sessions
G729r8. Additionally their CUCM will require 200 S/W MTP 4321 4 100
sessions.
4331 8 500
• 800 IP telephony sessions = +800 IPT calls 4351 10 1000
• 100 Contact Center calls = +400 IPT calls 4431 15 3000
• A call that needs to be recorded = an IPT call, +100 IPT calls 4451 40 6000
(Call Recording)
TOTAL = 1500 CUBE sessions
• PVDM4-128 for transcoding
But CPS expected is 20
• 1 S/W MTP session ~ 1 CUBE IPT session, +200 IPT calls (S/W
MTP sessions) Deploy a 4451 or two
4351s w/CUSP
CUBE Licensing Plans
NOTE : Subject to Change

CUBE Licensing
• Will still remain honor based
• July 2018 release (16.9.1) default is Classic Licensing mode (RTU)
• CUBE version 12.5 or later: (IOS-XE 16.10) Smart Licensing mode is the only option,
though CUBE feature use is still RTU (Right-to-Use). So even though the box will be
registered to CSSM, it won't demand any CUBE licenses. It will however, request UCK9,
security etc. and hence, it is recommended to familiarize yourselves with Smart
Licensing
• Note: Recommendation is to use IOS-XE 16.10.2 or later for CUBE 12.5 features
• Additional changes wrt to SL will be added for CUBE, however, there are no plans for
CUBE licensing enforcement, i.e. calls will not be denied should CSSM respond out of
compliance.
Agenda
CUBE Call Flow
CUBE Call Processing
CUBE
§ Actively involved in the call treatment, signalling

and media streams
IP
§ SIP B2B User Agent
Media Flow-Through
§ Signalling is terminated, interpreted and re-
§ Signaling and media terminated by the Cisco
originated Unified Border Element
§ Provides full inspection of signalling, and protection § Transcoding and complete IP address hiding
against malformed and malicious packets require this model
§ Media is handled in two different modes: CUBE
§ Media Flow-Through
IP
§ Media Flow-Around
Media Flow-Around
§ Digital Signal Processors (DSPs) are only required § Only Signaling is terminated on CUBE
for transcoding (calls with dissimilar codecs) § Media bypasses the Cisco Unified Border
Element
Cisco Unified Border Element Basic Call Flow
voice service voip
mode border-element
allow-connections h323 to h323
Originating allow-connections h323 to sip Terminating
allow-connections sip to h323
Endpoint - allow-connections sip to sip
Endpoint –
1000 2000
Incoming VoIP Call Outgoing VoIP Call
CUBE
dial-peer voice 1 voip dial-peer voice 2 voip

destination-pattern 1000 destination-pattern 2000
session protocol sipv2 session protocol sipv2
session target ipv4:1.1.1.1 session target ipv4:2.2.2.2
codec g711ulaw codec g711ulaw
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
Understanding the Call flow
Incoming VoIP Call Leg Outgoing VoIP Call Leg
Matches an Incoming Dial-peer Matches an Outbound Dial-peer
VRF1 – 10.10.10.10 CUBE 20.20.20.20 – VRF2

1000 2000
1.1.1.1 INVITE /w SDP INVITE /w SDP 2.2.2.2
c= 1.1.1.1
c= 20.20.20.20
m=audio abc RTP/AVP 0
m=audio xxx RTP/AVP 0
100 TRYING 100 TRYING
180 RINGING 180 RINGING

200 OK
200 OK c= 2.2.2.2
c= 10.10.10.10 m=audio uvw RTP/AVP 0
m=audio xyz RTP/AVP 0 ACK
ACK
RTP (Audio)
1.1.1.1 10.10.10.10 20.20.20.20 2.2.2.2
BYE BYE
200 OK
200 OK
Basic Show Commands for Active Calls
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
#CLUS LTRCOL-2310 © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 23
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
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
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
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
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
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
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
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
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
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
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
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
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
CUBE Architecture
ISR G2 vs ASR1K vs ISR 4K vs
vCUBE (CUBE on CSR)
ASR/ISR-4K & ISR-G2 Architecture Comparison
ASR/ISR-4K (IOS-XE) Architecture ISR G2 Architecture
CPU
IOS
RP Control Plane Control
Plane
IOS-XE IOS-XE
Signaling
I/O I/O
Signaling
Kernel Data Plane
§ ISR: Pkt fwd’ing and signaling are handled by the
Msg I/f same CPU
§ ASR: Pkt fwd’ing and signaling are handled by
different CPUs
I/O ESP I/O
‒ ESP must be programmed or instructed by the
control plane to do specific media functions
Data (Forwarding) Plane
‒ Performed by Forwarding Plane Interface (FPI)
Media
Virtual CUBE (CUBE on CSR 1000v) 5
3.1
Architecture XE
• CSR (Cloud Services Router) 1000v runs on a Hypervisor – IOS

XE without the router
ESXi Container
RP (control plane) ESP (data plane) FFP code
Chassis Mgr. QFP Client Chassis Mgr.

IOS-XE Forwarding Mgr. / Driver Forwarding Mgr.
CUBE signaling CUBE media processing

Kernel (incl. utilities)
Virtual CPU Memory Flash / Disk Console Mgmt ENET Ethernet NICs
CSR 1000v (virtual IOS-XE)
Hypervisor vSwitch NIC
X86 Multi-Core CPU Memory Banks Hardware GE … GE

Virtual CUBE (CUBE on CSR 1000v) – Cont’d
5
3.1
XE
• CSR1000v is a virtual machine, running on x86 server (no specialized 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
• vCUBE tracks only the next vSwitch interface resulting in SSO of vCUBE-HA only due to software
failures (active vCUBE crashing/reloading)
• 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
#CiscoLiveLA © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public
Agenda
Step 1:
Configure CUCM to route calls to the edge SBC
SIP Trunk Pointing to CUBE
Standby
CUBE
A
Active IP PSTN
CUBE
Enterprise CUBE with High
Campus Availability
MPLS
• Configure CUCM to route all PSTN
PSTN is now
calls (central and branch) to CUBE
used only for
(Gig0/0
SRST
in our slides) via a SIP trunk
emergency
calls over
FXO lines
• Make sure all different patterns of
calls – local,
CME long distance,
international, emergency,TDM PBX
informational
Enterprise etc.. are pointing to
Branch Offices
CUBE #CiscoLiveLA BRKCOL-2125 © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 42
Step 2: Get details from SIP Trunk provider
Item SIP Trunk service provider requirement Sample Response
1 SIP Trunk IP Address (Destination IP Address for INVITES) 66.77.37.2 or DNS
2 SIP Trunk Port number (Destination port number for INVITES) 5060
3 SIP Trunk Transport Layer (UDP or TCP) UDP
4 Codecs supported G711, G729
5 Fax protocol support T.38
6 DTMF signaling mechanism RFC2833
7 Does the provider require SDP information in initial INVITE (Early offer Yes
required)
8 SBC’s external IP address that is required for the SP to
128.107.214.195
accept/authenticate calls (Source IP Address for INVITES)
9 Does SP require SIP Trunk registration for each DID? If yes, what is the No
username & password
10 Does SP require Digest Authentication? If yes, what is the username & 408-944-7700
password#CiscoLiveLA © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public
Step 3: Enable CUBE Application on Cisco routers
1. Enable CUBE Application
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 (CPU intensive CLI)
sip
early-offer forced
3. Create a trusted list of IP addresses to prevent toll-fraud
voice service voip
ip address trusted list à Applications initiating signaling 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 ! 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
Step 4: Configure Call routing on CUBE
10.10.1.20
66.77.37.2
10.10.1.21 128.107.214.195
• 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 call legs to and
from the PBX. Always bind LAN interface(s) on CUBE to LAN dial-peers, ensuring SIP/RTP is sourced
from the intended LAN interfaces(s)
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for sending and receiving
call legs to and from the ITSP. Always bind CUBE’s WAN interface(s) to WAN dial-peer(s).
WAN Dial-Peer Configuration
Inbound Dial-Peer for call legs from SP to CUBE
dial-peer voice 200 voip
description *** Inbound WAN side dial-peer *** Specific to your DID range
incoming called-number +1408944 . . . . $
assigned by the SP
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
Apply bind to all dial-peers when
codec g711ulaw CUBE has multiple interfaces.
dtmf-relay rtp-nte Gig0/1 faces SP.
no vad
Outbound Dial-Peer for call legs from CUBE to SP
dial-peer voice 201 voip Translation rule/profile to strip the
description *** Outbound WAN side dial-peer *** access code (9) before delivering
translation-profile outgoing Digitstrip the call to the SP
destination-pattern 91[2-9]..[2-9]......$
session protocol sipv2 Dial-peer for making long distance
voice-class sip bind control source gig0/1 calls to SP, based on NANP (North
American Numbering Plan)
session target ipv4:66.77.37.2 ! ITSP SIP Trunk
codec g711ulaw Note: Separate outgoing DP to be created for Local, International,
dtmf-relay rtp-nte Emergency, Informational calls etc.
no vad #CiscoLiveLA 46
BRKCOL-2125 © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public
LAN Dial-Peer Configuration
Inbound Dial-Peer for call legs from CUCM to CUBE
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
CUCM sending 9 (access code) + All
session protocol sipv2 digits dialed
Apply bind to all dial-peers when
codec g711ulaw CUBE has multiple interfaces. Gig0/0
dtmf-relay rtp-nte faces CUCM.
no vad
Outbound Dial-Peer for call legs from CUBE to CUCM
description *** Outbound LAN side dial-peer *** SP will be sending 10 digits (NANP)
destination-pattern +1408944 . . . . $ based on your DID that is being
session protocol sipv2 delivered to CUCM
session target ipv4:10.10.1.20 ! CUCM IP Address
codec g711ulaw Default codec is G729 if none is
dtmf-relay rtp-nte specified
no vad
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
SIP Normalization t ing Toolort/web/
rof sco.com/c/en ile tes /us/supp
SIP profiles is a mechanism to normalise or customise SIP at the SIP /P
/www.
ci
:
network border to provide interop between incompatible devices https catalog.html
tools-
SIP incompatibilities arise due to: Add user=phone for INVITEs
• A device rejecting an unknown header (value or Incoming Outgoing
parameter) instead of ignoring it CUBE
INVITE INVITE
sip:5551000@sip.com:5060 sip:5551000@sip.com:5060
• A device expecting an optional header SIP/2.0 user=phone SIP/2.0
value/parameter or can be implemented in
voice class sip-profiles 100
multiple ways 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"
• A device sending a value/parameter that must be
changed or suppressed (“normalised”) before it Modify a sip: URI to a tel: URI in INVITEs
leaves/enters the enterprise to comply with
policies Incoming Outgoing
CUBE
INVITE INVITE
• Variations in the SIP standards of how to achieve sip:2222000020@9.13.24.6:5060 tel:2222000020
certain functions SIP/2.0 SIP/2.0

• With CUBE 10.0.1 SIP Profiles can be request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1"
request INVITE sip-header From modify "<sip:(.*)@.*>" "<tel:\1>"
applied to inbound SIP messages as request INVITE sip-header To modify "<sip:(.*)@.*>" "<tel:\1>"
well
More information at http://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/118825-technote-sip-00.html
SIP Profile Rule Tagging
SIP Profile Tagging Enhancement . 5.2T
15 3.15
/
New rule tagging mechanism is being introduced XE
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.
T/
SIP Profile Tagging – Configuration 5.2
15. 3.15
XE
• For tagging the rules, an additional option of “rule <tag>” has been provided
CUBE(config)#voice class sip-profiles 1
CUBE(config-class)#?
VOICECLASS configuration commands: The new
exit Exit from voice class configuration mode keyword “rule”
help Description of the interactive help system
no Negate a command or set its defaults
request sip request
response sip response “tag” to be
rule Specify the rule provided with
rule keyword
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
response sip response
SIP Profile Tagging – Configuration Cont’d 5.
1 3.1
XE
T/
5.2 5
• For inserting a rule between two rules, “before” option has been provided
CUBE(config)#voice class sip-profiles 1 For inserting a rule

CUBE(config-class)#rule before ?
between two rules, the
<1-1073741823> Specify the rule tag
new before keyword
CUBE(config-class)#rule before 3 ? is being introduced
request sip request
response sip response
• 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>
Configuration Example 5.
1 3.1
T/
5.2 5
XE
• 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”

before
voice class sip-profiles 1 option
rule 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”
rule 3 request INVITE sip-header Supported Add “Supported: ” The new rule has
been inserted
between #1 and #3
Configuration Example Continued…. 5.
1 3.1
T/
5.2 5
XE
• 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”
Configuration Example Continued…. 5.
T/
5.2 5
1 3.1
XE
• 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”
SIP Profile Support for Non-
Standard Headers
SIP Profile Support for Non-Standard Headers . 5.2T
15 3.15
/
XE
§ 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 The new “WORD”
……. option for specifying
Via SIP header Via unsupported headers
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”
Agenda
CUBE Dial-Peers
Call Routing
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. Should be bound to the LAN interface(s) of CUBE to ensure
SIP/RTP is sourced from the LAN IP(s) of the CUBE.
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for sending &
receiving calls to & from the provider. Should be bound to WAN interface(s) of CUBE.
Inbound LAN Dial-Peer Outbound Calls Outbound WAN Dial-Peer
A
CUCM SIP Trunk ITSP SIP Trunk
IP PSTN
CUBE
Inbound Calls
Outbound LAN Dial-Peer Inbound WAN Dial-Peer
Understanding Inbound Dial-Peer Matching Techniques
Priority
Inbound LAN Dial-Peer Outbound Calls
Exact Pattern
Match Based on URI match A CUCM SIP Trunk SP SIP Trunk
IP
1 of an incoming Host Name/IP CUBE PSTN
INVITE message Address
Inbound Calls
Inbound WAN Dial-Peer
User portion of
2 Match based on URI Received:
Called Number Phone-number of INVITE sip:654321@10.2.1.1 SIP/2.0
tel-uri Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
3 Match based on From: "555" <sip:555@10.1.1.1:5060>;tag=1
Calling number To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
4 Default Dial-Peer = 0 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
........
Priority Inbound LAN Dial-Peer Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1 A CUCM SIP Trunk SP SIP Trunk IP
voice class uri 2001 sip PSTN
host ipv4:10.2.1.1 CUBE
A Inbound Calls
incoming uri via 1001 Inbound WAN Dial-Peer
1 B
dial-peer voice 2 voip Received:
incoming uri request 2001 INVITE sip:654321@10.2.1.1 SIP/2.0
C dial-peer voice 3 voip Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
incoming uri to 2001 tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
D dial-peer voice 4 voip
To: ABC <sip:654321@10.2.1.1:5060>
incoming uri from 1001
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
dial-peer voice 5 voip Contact: sip:555@10.1.1.1:5060
2 incoming called-number 654321 Supported: timer
Max-Forwards: 70
dial-peer voice 6 voip Subject: BRKUCC-2934 Session
3 answer-address 555 Content-Type: application/sdp
Content-Length: 226
4 dial-peer voice 7 voip
destination-pattern 555
........
Priority
voice class uri 1001 sip Inbound LAN Dial-Peer Outbound Calls
host ipv4:10.1.1.1
voice class uri 2001 sip A CUCM SIP Trunk SP SIP Trunk IP
PSTN
A dial-peer voice 1 voip Inbound Calls
incoming uri via 1001
1 B dial-peer voice 2 voip Received:
incoming uri request 2001
C INVITE sip:654321@10.2.1.1 SIP/2.0
dial-peer voice 3 voip Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
D dial-peer voice 4 voip From: "555" <sip:555@10.1.1.1:5060>;tag=1
incoming uri from 1001 To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
2 incoming called-number 654321
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
3 answer-address 555
Content-Length: 226
........
host ipv4:10.1.1.1
A CUCM SIP Trunk SP SIP Trunk IP
PSTN
1 B dial-peer voice 2 voip
incoming uri request 2001 Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
C dial-peer voice 3 voip Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
From: "555" <sip:555@10.1.1.1:5060>;tag=1
D dial-peer voice 4 voip
incoming uri from 1001
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
3 answer-address 555
Content-Length: 226
........

A CUCM SIP Trunk SP SIP Trunk IP
host ipv4:10.1.1.1
PSTN
voice class uri 2001 sip CUBE
host ipv4:10.2.1.1 Inbound Calls
A
dial-peer voice 1 voip Inbound WAN Dial-Peer
1 B dial-peer voice 2 voip Received:
incoming uri request 2001 INVITE sip:654321@10.2.1.1 SIP/2.0
C dial-peer voice 3 voip
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
From: "555" <sip:555@10.1.1.1:5060>;tag=1
D dial-peer voice 4 voip To: ABC <sip:654321@10.2.1.1:5060>
incoming uri from 1001 Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
incoming called-number 654321 Supported: timer
Max-Forwards: 70
answer-address 555 Content-Type: application/sdp
Content-Length: 226
4 dial-peer voice 7 voip ........

dial-peer voice 3 voip INVITE sip:654321@10.2.1.1 SIP/2.0
C incoming uri to 2001 Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
dial-peer voice 4 voip tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
D incoming uri from 1001 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
dial-peer voice 5 voip CSeq: 1 INVITE
2 incoming called-number 654321 Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
answer-address 555
Content-Length: 226
4 dial-peer voice 7 voip ........
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
dial-peer voice 5 voip CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
dial-peer voice 6 voip Max-Forwards: 70
3 answer-address 555 Subject: BRKUCC-2934 Session
Content-Length: 226
........
CUBE
host ipv4:10.2.1.1
A Inbound Calls
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
dial-peer voice 6 voip Supported: timer
3 answer-address 555 Max-Forwards: 70
4 dial-peer voice 7 voip Content-Length: 226
destination-pattern 555 ........
Outbound Dial-Peer Matching Criteria Summary
Priority Outbound WAN Dial-Peer
Outbound Calls
Match Based on DPG,
0 DPPP, COR/LPCOR if A CUCM SIP Trunk SP SIP Trunk
IP
configured CUBE PSTN
Exact Pattern Outbound LAN Inbound Calls
match Dial-Peer
Match Based on Host Name/IP Received:
1 URI of incoming Address INVITE sip:654321@10.2.1.1 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
INVITE message User portion of URI tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
Phone-number of To: ABC <sip:654321@10.2.1.1:5060>
tel-uri Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Match based on Contact: sip:555@10.1.1.1:5060
2 Called Number Supported: timer
Max-Forwards: 70
Content-Length: 226
........
description *Outbound WAN dial-peer. From CUBE to SP*
dial-peer voice 100 voip destination-pattern 91[2-9]..[2-9]......$
description *Inbound LAN dial-peer. From CUCM to CUBE* session protocol sipv2
session protocol sipv2 session target ipv4:66.77.37.2
incoming called-number 9T
voice-class sip bind control source-interface Gig0/0 session transport udp
voice-class sip bind media source-interface Gig0/0 voice-class sip bind control source-interface Gig0/1
dtmf-relay rtp-nte voice-class sip bind media source-interface Gig0/1
codec g711ulaw dtmf-relay rtp-nte
no vad codec g711ulaw
no vad
Inbound LAN Dial-Peer Outbound WAN Dial-Peer
Outbound Calls
A
CUCM SIP Trunk ITSP SIP Trunk
G0/0 - 10.10.1.21 CUBE G0/1 – 128.107.214.195
66.77.37.2
10.10.1.20
Outbound LAN Dial-Peer Inbound Calls Inbound WAN Dial-

dial-peer voice 200 voip Peer
dial-peer voice 101 voip description *Inbound WAN dial-peer. From Provider to CUBE*
description *Outbound LAN dial-peer. From CUBE to CUCM* session protocol sipv2
translation-profile outgoing CUBE_to_CUCM incoming uri via 200
destination-pattern +1408944....$ voice-class sip bind control source-interface Gig0/1
session protocol sipv2 voice-class sip bind media source-interface Gig0/1
session target ipv4:10.10.1.20 dtmf-relay rtp-nte
voice-class sip bind control source-interface Gig0/0 codec g711ulaw
voice-class sip bind media source-interface Gig0/0 no vad
dtmf-relay rtp-nte
codec g711ulaw voice class uri 200 sip
no vad #CiscoLiveLA © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public
host ipv4:66.77.37.2
OPTIONS KeepAlive
CUBE SIP Trunk Monitoring with OOD Options message
• Out-of-dialog OPTIONS message sent
ITSP SIP Trunk
A CUCM SIP Trunk ITSP to check the status of the SIP Trunk
on an Outbound LAN/WAN dial-peer
CUBE
OOD Options
200 OK • The dial-peer is “busyout” if it does

Dial-Peer 201 = ACTIVE not receive a response within a
INVITE
INVITE configurable time period or receives
200 OK
200 OK
a 503 or 505
OOD Options
• For an INVITE that matches a
Timeout – no
response
“busyout” dial-peer, CUBE sends
Dial-peer 201 = BUSYOUT “503 Service Unavailable”
INVITE OOD Options
• If there is a secondary dial-peer
503 Service Unavailable
OOD Options configured, the call will be re-routed
the secondary path
CUBE SIP Trunk Monitoring with OOD Options message
A CUCM SIP Trunk SP SIP Trunk voice-class sip options-
SP
CUBE SIP keepalive up-interval 20 down-
OOD Options interval 20 retry 3
200 OK
Three timers that can be configured:
DP 100 =
INVITE ACTIVE • up-Interval: OPTIONS keepalive timer
INVITE
200 OK
interval for UP endpoint
200 OK
• down-interval: OPTIONS keepalive timer
OOD Options interval for DOWN endpoint
• retry: Retry count for OPTIONS keepalive
Timeout – no
response transmission
DP 100 = BUSYOUT
Warning:
INVITE OOD Options
• Each dial-peer that has options message
503 Service Unavailable
configured sends out a separate message.
OOD Options
• EEM Script can be used to busyout other
dial-peers
OOD OPTIONS Ping Keepalive Enhancement .4.1T
15 3.11
/
XE
• Each dial-peer that has OPTIONS message configured

A
CUCM SIP Trunk SP SIP Trunk
SP SIP
sends out a separate message, even if the session
CUBE targets are same
OOD Options (DP 201)
200 OK • Network bandwidth and process runtime are wasted in
DP 201 : Session Target IPv4:66.77.37.2 CUBE and remote targets to sustain duplicate OOD
OPTIONS Ping heartbeat keepalive connection
INVITE INVITE (DP 100) • Consolidate SIP OOD Options Ping connections by
200 OK
200 OK grouping SIP dial-peers with same OOD Options Ping
setup
200 OK • New CLI : “voice class sip-keepalive-
DP 211: Session Target IPv4:66.77.37.2 profile <tag>” is used to define OOD OPTIONS Ping
setup
200 OK
• Consolidated SIP OOD Options Ping connection will then
be established with a target for multiple SIP dial-peers
DP 400: Session Target IPv4:1.1.1.1
with the same target and OOD Options Ping profile
setup
OOD OPTIONS Ping Keepalive Enhancement –
Configuration
voice class sip-options-keepalive 1 Sample Show command output
description UDP Options consolidation CUBE#sh voice class sip-options-keepalive 1
down-interval 49 Voice class sip-options-keepalive: 1 AdminStat: Up
up-interval 180 Description: UDP Options consolidation
retry 7 Single OOD Option
Ping Group applied Transport: udp Sip Profiles: 0
transport udp
to multiple dial-peers Interval(seconds) Up: 180 Down: 49
with same session
dial-peer voice 201 voip targets Retry: 7
session protocol sipv2 Peer Tag Server Group OOD SessID OOD Stat IfIndex
session target ipv4:10.104.45.253 -------- ------------ ---------- -------- -------
voice-class sip options-keepalive profile 1
201 4 Active 9
211 4 Active 10
OOD SessID: 4 OOD Stat: Active
session protocol sipv2 Target: ipv4:10.104.45.253
session target ipv4:10.104.45.253 Transport: udp Sip Profiles: 0
voice-class sip options-keepalive profile 1
• With OOD Options Ping Keepalive group, an options ping keepalive connection is established on per remote target base as opposed an options ping keepalive connection
established per dial-peer basis. Up to 10,000 “voice class sip-options-keepalive <tag>” can be defined per system
• Either legacy “sip options-keepalive” or the new “sip options-keepalive profile <tag>” can be configured on a dial-peer. Dial-peers with Destination Server Group instead of
Session Target IP must use Options Keepalive Profile and not the legacy CLI.
T/
4.1 1
Destination Server Group 5.
1 3.1
XE
• 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
voice class server-group 1 dial-peer voice 100 voip

hunt-scheme {preference | round-robin} description Outbound DP
ipv4 1.1.1.1 preference 5 destination-pattern 1234
ipv4 2.2.2.2 session protocol sipv2
ipv4 3.3.3.3 port 3333 preference 3 codec g711ulaw
ipv6 2010:AB8:0:2::1 port 2323 preference 3 dtmf-relay rtp-nte
ipv6 2010:AB8:0:2::2 port 2222 session server-group 1
* DNS target not supported in server group
Multiple Incoming Patterns Under Same 5.
T/
4.1 1
Incoming/Outgoing Dial-peer
1 3.1
XE
voice class e164-pattern-map 300 Provides the ability to combine multiple

e164 919200200.
incoming called OR calling numbers on a
e164 510100100.
Site A (919)200-2000
e164 408100100. single inbound voip dial-peer, reducing
the total number of inbound voip dial-
Site B (510)100-1000
description Inbound DP via Calling peers required with the same routing
incoming calling e164-pattern-map 300 capability
Site C (408)100-1000 codec g729r8 Up to 5000 entries in a text file
G729 Sites A SIP Trunk SP SIP Trunk

IP PSTN
CUBE
Site A voice class e164-pattern-map 400 ! This is an example of the contents of

(919)200-2010
url flash:e164-pattern-map.cfg E164 patterns text file stored in
flash:e164-pattern-map.cfg
Site B (510)100-1010dial-peer voice 2 voip
description Outbound DP via Called 9192002010
Site C (408)100-1010 5101001010
destination e164-pattern-map 400
codec g711ulaw 4081001010
G711 Sites
T/
. 4.1 1
Destination Dial-Peer Group 5
1 3.1
XE
• 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
T/
Destination Dial-peer Group Configuration 5. 4.1 1
1 3.1
XE
voice class dpg 10000 dial-peer voice 1001 voip

description Voice Class DPG for SJ destination-pattern 2222
dial-peer 1001 preference 1 session protocol sipv2
dial-peer 1002 preference 2 session target ipv4:10.1.1.1
3.
dial-peer 1003 DP O u t !
is s b ou
! e le n d dial-peer voice 1002 voip
cte
dial-peer voice 100 voip d destination-pattern 3333
description Inbound DP session protocol sipv2
incoming called-number 1341 session target ipv4:10.1.1.2
destination dpg 10000 !
session target ipv4:10.1.1.3
1. Incoming Dial-peer is first
matched 2. Now the DPG associated with
the INBOUND DP is selected
Media Manipulation
Understanding the difference between MTPs and Transcoders
• Only a CUCM requirement as a placeholder, not needed
by IOS/IOS-XE devices. Always invoked by CUCM
HEADER MTP
RTP packet
• Most commonly used for DTMF interworking, IPv4-IPv6

TRANSCODER (universal, regular)
PAYLOAD • Always requires DSPs hosted in an ISR/ASR
• Can be invoked by CUCM (SCCP) or CUBE (LTI –
DSPs must be in the same box as CUBE)
H/W Uses
Only G711
IOS/IOS- based DSPs
XE based
S/W CPU Exception is Codec
based Any Codec Passthru, e.g T38
MTPs cycles
CUCM CPU Exception : Can

G711 or convert from alaw to
based Cycles G729 (v9+) ulaw, vice versa
Audio Transcoding and Transrating
iLBC, iSAC,
Speex Enterprise IP Phones:
SP VoIP VoIP G.711, G.729 20 ms,
CUBE G.722
G.729 30 ms
• Transcoding (12.4.20T) • Transcoding: G.711, G.723.1, G.726, G.728,

G.729/a, iLBC, G.722
• One voice codec to any other codec E.g. iLBC-G.711 or iLBC- • Transrating: G.729 20ms ↔ 30ms (AT&T)
G.729
• CUCM 7.1.5 or later supports universal Transcoding Packetization
Supported Codecs
(ms)
• Transrating (15.0.1M) G.711 a-law 64 Kbps 10, 20, 30
• Different packetizations of the same codec
G.711 µlaw 64 Kbps 10, 20, 30
• E.g. G.729 20ms to G.729 30ms
• Support for SIP-SIP calls G.723 5.3/6.3 Kbps 30, 60
• No sRTP support with transrating
G.729, G.729A, G.729B, 10, 20, 30, 40, 50,
dial-peer voice 2 voip !Call volume (gain/loss) adjustment G.729AB 8 Kbps 60
codec g729r8 bytes 30 fixed-bytes dial-peer voice 2 voip
audio incoming level-adjustment x G.722—64 Kbps 10, 20, 30
audio outgoing level-adjustment y
Configuration for SCCP based Transcoding Y
r
ou ce
n
(ISR-G2/4K) r
Fo fere
Re
1. Enabling dspfarm services 3. SCCP configuration
under voice-card
sccp local GigabitEthernet0/0
voice-card 1 sccp ccm <CUBE_internal_IP> identifier 1 version 7+
dspfarm ! Only ISR G2
dsp services dspfarm sccp
sccp ccm group 1
2. telephony-service configuration associate ccm 1 priority 1
associate profile 1 register CUBE-XCODE
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 128 4. dspfarm profile configuration
sdspfarm tag 1 CUBE-XCODE
max-ephones 10 dspfarm profile 1 transcode
codec g711ulaw
max-dn 10
ip source-address codec g711alaw
<CUBE_internal_IP> port 2000 codec g729r8
maximum sessions 10
associate application SCCP
Configuration for LTI based Transcoding
(ISR-G2/4K & ASR)
1. Enabling dspfarm services
under voice-card
voice-card 0/1 Feature Notes:
dspfarm ! Only ISR G2 • This uses Local Transcoding Interface to
dsp services dspfarm
communicate between CUBE and DSPs
2. dspfarm profile configuration • Also available on ISR-G2 starting IOS 15.2.3T
• Can only be used if CUBE invokes the DSP
dspfarm profile 1 transcode for media services
codec g711ulaw
codec g711alaw • CUCM cannot invoke DSPs using this LTI
codec g729abr8 interface
codec g729ar8 • Supported with CUBE HA
codec ilbc
maximum sessions 100
associate application CUBE
Serviceability
/
.1T
New CUBE Serviceability Features . 3
15 3.8
XE
Call Arrival Rate
Example:
show call history stats cps
Histogram for Call rate 1122222357676678753222211111122247545789774322213311112245654598843333222

10
Histogram for Concurrent calls 9 * *
8 * ** ***
Histogram for Call duration 7 * * *** * ***** * ##*
6 ******** * ***** ** *##*
Histogram for SIP message rate 5 *########* #* *####* *######*
4 *########* *#***####** *########*
High/Low watermark for Call Rate
3 **########** *#########** ** *########*****
High/Low watermark for Concurrent calls 2 ******#########***** ****##########**** ** ***########********
1 *######################################################################*
0....5....1....1....2....2....3....3....4....4....5....5....6....6....7..
High/Low watermark for SIP message rate 0 5 0 5 0 5 0 5 0 5 0 5 0
Call switching rate / CPS (last 72 hours)
* = maximum calls/s # = average calls/s
Histogram for Call Failure Rate
High/Low watermark for Call Failure Rate
Call History Stats – Graphical or Tabular form
Last 60 sec, 60 minutes, 72 hours
show call history stats connected [table]
Ability to sort dial-peers 5.
1 3.1
T/
5.2 5
show run dial-peer sort XE
dial-peer (default) dial-peer sort dial-peer sort descending

dial-peer voice 4020 pots dial-peer voice 5 pots dial-peer voice 5000 voip
destination-pattern 4020 incoming called-number 1... destination-pattern 5...
port 0/2/0 port 1/0/0:23 session protocol sipv2
! ! session target ipv4:1.4.65.5
dial-peer voice 5000 voip dial-peer voice 4020 pots !
destination-pattern 5... destination-pattern 4020 dial-peer voice 4020 pots
session protocol sipv2 port 0/2/0 destination-pattern 4020
session target ipv4:1.4.65.5 ! port 0/2/0
Dial Peer tag
! dial-peer voice 5000 voip !
dial-peer voice 5 pots destination-pattern 5... dial-peer voice 5 pots
incoming called-number 1... session protocol sipv2 incoming called-number 1...
port 1/0/0:23 session target ipv4:1.4.65.5 port 1/0/0:23
Total Number of Active Concurrent Calls . 3
15 3.10
1
.3M .1
Total Number of Active Calls XE
Router# show call active total-calls

§ A single call can have multiple call-legs.
To determine the total number of Total Number of Active Calls : 10
active calls from call-legs is challenging Connected
Call Flow Call-legs
§ CLI added to display the value of call
current number of active (connected) Basic call (audio/video) 2 1
calls on CUBE
Transferred call (Refer handling) 3 2
§ The table defines the relation between
call-legs and number of active calls
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 (signaling forking) 2 1
Avoiding Non-Call-Context Debug Logs . 4.2T
15 3.12
/
XE
• 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
#CiscoLiveLA BRKUCC-2006 © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 90
Debugging Made Easier . 3
15 3.10
1
.3M .1
XE
Categorize Debugs based on Functionality
§ Categorization based on
Functionality Router# debug ccsip feature < audio | cac |
1. Audio/video/sdp/control config | control | dtmf | fax | line | misc |
2. Configuration /sip-transport misc-features | parse | registration | sdp-
3. CAC negotiation | sdp-passthrough | sip-profiles |
4. DTMF/FAX/Line-side sip-transport | srtp | supplementary-services
5. Registration | transcoder | video >
6. Sdp - passthrough
7. Sip-profile/SRTP/transcoder
Example: enabling DTMF and audio debugs only with default log level is considered.
DTMF(32) debug code
CUBE#sh debugging
CCSIP SPI: SIP info debug tracing is enabled (filter is OFF)
CCSIP SPI: audio debugging for ccsip info is enabled (active) Audio(2) debug code
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:
1
.3M .1
Debugging Made Easier . 3
15 3.10
XE
Categorize Debugs based on Functionality|-----------------------------------------------
| show cube debug category codes values.
CUBE# show cube debug category codes |-----------------------------------------------
| Indx | Debug Name | Value
|-----------------------------------------------
| 01 | SDP Debugs | 1
§ This CLI is used to collect the | 02 | Audio Debugs | 2
predefined debug features category | 03 | Video Debugs | 4
| 04 | Fax Debugs | 8
codes , which helps in analysis of | 05 | SRTP Debugs | 16
debugs manually. | 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
|-----------------------------------------------
External/PSTN Call
Recording
External/PSTN Call Recording Options
• 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
T/
CUBE Controlled Recording Option - SIPREC 6.1
15. 3.17
XE
Dial-peer based – SIPREC Standard
• 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
Recorder • Metadata includes the
SIPREC Compliant Recorder
communication session details of
P audio or video calls and also
SI
P identifies the participants of the call
RT XML Metadata
A SIP SIP
SP SIP
CUBE
RTP RTP
media class 9 h
to matc dial-peer voice 950 voip
recorder parameter siprec N eed s
• SIP Profiles can description dial-peer pointing to MediaSense
additionally be used to media-recording 950 Needs to

destination-pattern 9999 ! Dummy
forward 3rd party IP PBX dial-peer voice 901 voip match
session transport tcp
Call Identifier to the description dial-peer that needs to be forked session target ipv4:<Recorder_IP>
Recorder for Correlation session protocol sipv2 ! Bind on this DP mandatory
media-class 9
Sample SIPREC XML Metadata Generated
CUCM (10.X or later) Controlled Recording .3
1
.3M 1
15 .10. .0
UC Services API – Network Based Recording 3 0
XE C M 1
3. C U
1. Enable HTTP on IOS

Gateway/CUBE Recording ip http server
Enabled http client persistent
2. Enable the API on IOS

4. uc wsapi
1. 2. source-address [IP_Address_of_CUBE]
3. Enable XMF service within the API

5. provider xmf
remote-url 1 http://CUCM:8090/ucm_xmf
no shutdown
[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
• 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] #CiscoLiveLA 97
BRKCOL-2125 © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multiple Non-
Authenticated SIP Trunks
on a CUBE
Non-Authenticated SIP Trunking to more than one
Service Provider
SIP SP-1 Large enterprises are deploying more
(10.10.10.2)
than one SIP Trunk provider for:
A
Active • Alternate call routing
CUBE SIP SP-2 • Load balancing
(20.20.20.2)
Enterprise
Campus
MPLS
interface loopback1 dial-peer voice 20 voip
SIP SP-1’s ip address 10.10.10.1 255.255.255.0 description “Secondary path to SIP SP-2”
network destination-pattern 91[2-9]..[2-9]......
interface loopback2 session protocol sipv2
SIP SP-2’s ip address 20.20.20.1 255.255.255.0 session target ipv4:20.20.20.2
SRST
network preference 2
dial-peer voice 10 voip voice-class sip options-keepalive
description “Primary path to SIP SP-1” voice-class sip bind control source-interface loopback2
destination-pattern
CME 91[2-9]..[2-9]...... voice-class sip bind media source-interface loopback2
session target ipv4:10.10.10.2 TDM PBX NOTE: Dual SPs can be used for outbound calls, but to
Enterprisesip options-keepalive
voice-class be utilised for inbound calls, arrangements between
Branch Offices
voice-class sip bind control source-interface loopback1 SPs required
voice-class sip bind media source-interface loopback1
Multiple Authenticated /
Registered SIP Trunks on a
CUBE
Multiple Instances of SIP-UA on a CUBE
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
/
.6.2T .1
15 16.3
XE
• 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 15.
T/
6.2 .1
16.
3
XE
• 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
Prior to Multi Tenancy
sip-ua
registrar 1 ipv4:60.60.60.60:9051 expires 3600 Global OB Proxy and Bind
registrar 2 ipv4:70.70.70.70:9052 expires 3600
credentials username aaaa password 7 06070E204D realm aaaa.com
credentials username bbbb password 7 110B1B0715 realm bbbb.com E164 - aaaa Registrar - 1
voice service voip
outbound-proxy ipv4:10.64.86.35:9057
bind control source-interface GigabitEthernet0/1 E164 - bbbb Registrar - 2
With Voice Class Tenant (Multi-Tenancy)

voice class tenant 1
registrar 1 ipv4:60.60.60.60:9051 expires 3600 OB Proxy 1 & Bind-1
credentials username aaaa password 7 06070E204D realm aaaa.com E164 - aaaa Registrar - 1
bind control source-interface GigabitEthernet0/0
voice class tenant 2
registrar 1 ipv4:70.70.70.70:9052 expires 3600
OB Proxy 2 & Bind-2
credentials username bbbb password 7 110B1B0715 realm bbbb.com E164 - bbbb Registrar - 1
Configuring Voice Class Tenant
• Configure voice class tenant
voice class tenant 1 Add new voice class
registrar 1 ipv4:10.64.86.35:9052 expires 3600 tenant
credentials username aaaa password 7 06070E204D realm aaaa.com
credentials number bbbb username bbbb password 7 110B1B0715 realm bbbb.com
bind media source-interface GigabitEthernet0/0 )
onfigs
lc
rder
copy-list 1
O gl oba
e d
fe r enc an t an )
early-offer forced n P re s ten
l c onfig
io e )
fi gurat (o v errid e globa s i p-ua
Con eer h OR
• Apply tenant to the desired dial-peer al -p rri des t v oi p
di e e
1. an t (ov s ervic
ten v oice
dial-peer voice 1 voip 2. l (
destination-pattern 111 g loba
3.
session target ipv4:10.64.86.35:9051
session transport udp Apply Tenant to a Dial-
voice-class sip tenant 1 peer
High Availability Updates
CUBE HA Design Considerations on ASR1K/ISR-4K/vCUBE for Box-
to-Box Redundancy
• 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
• A max of 2 CUBE HA pairs can be deployed in a single layer 2 domain (RG Control/Data interface)
• Starting IOS-XE 16.3.1, Port channel is supported for both RG Control/data and traffic interfaces
• All signaling/media is sourced from/to the Virtual IP Address
CUBE HA Design Considerations on ASR1K/ISR-4K/vCUBE for Box-
to-Box Redundancy
• Anytime a platform is reloaded in a CUBE-HA relationship, it always boots up as Standby

• Lower address for all the interfaces (GE0/0/0, GE0/0/1, and GE0/0/2) should be on the same platform
• Redundancy Interface Identifier, rii (HSRP Group number) should be unique to a pair/interface
combination on the same L2
• Configuration on both the CUBEs must be identical including physical configuration and must be
running on the same type of platform and IOS-XE version. Loopback interfaces cannot be used as
bind as they are always up. Sub-interfaces are supported
• Multiple RII Groups/Interfaces/sub-interfaces can be used on either LAN or WAN side
CUBE HA Design Considerations on ASR1K/ISR-4K/vCUBE for Box-to-Box
Redundancy
• Multiple traffic (SIP/RTP) interfaces (GE0/0/1, GE0/0/1) require interface tracking to be configured
• Upto 6 multimedia lines in the SDP are checkpointed for CUBE HA. SDP Passthru (upto 2 m-lines) calls
are also checkpointed starting IOS-XE 3.17
• No media-flow around or UC Services API (CUCM NBR) support for CUBE HA
• CUBE-HA preserves both signaling and media and is not supported over a crossover cable connection
for the RG-control/data link (GE0/0/2)
• Both platforms must be connected via a physical Switch across all likewise interfaces for CUBE HA to
work, i.e. GE0/0/0 of CUBE-1 and CUBE-2 must terminate on the same switch and so on. Multiple
interfaces/sub-interfaces can be used on either LAN or WAN side
Redundancy
• Cannot have WAN terminated on CUBEs directly or Data HA on either side. Both Active/Standby must
be in the same Data Center
• CUBE HA only checkpoints SIP/RTP Traffic. Support for Survivability.tcl preservation was added in IOS-
XE 3.17 for CVP deployments
• CCB (courtesy callback) feature is not supported if a callback was registered with CVP and then a
switchover was done on CUBE. The CCB will not work in these scenarios.
Redundancy
• LTI based transcoding called flows are preserved starting IOS-XE 3.15. Requires same DSP module
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
• Upon failover, the previously ACTIVE CUBE goes through a reload by design, preserving
signaling/media
• Running configuration should always be saved to avoid losing it due to the reload by design
• It is mandatory to use separate L3 interface for redundancy (RG Control/data, GE0/0/2). i.e interface
used for traffic cannot be used for HA keepalives and checkpointing
• When configuration is applied and saved, the platform must go through a reload cycle
CUBE Configuration on ASR/ISR-4K/vCUBE Box-to-Box
Redundancy
CUBE 1 CUBE 2
Disables software redundancy redundancy
redundancy
mode none For ASR1006: mode rpr mode none
application redundancy application redundancy
group 1 group 1
name voice-b2bha Configure RG name voice-b2bha
priority 100 failover threshold 75 Group for use priority 100 failover threshold 75
control GigabitEthernet 0/0/2 protocol 1 control GigabitEthernet 0/0/2 protocol 1
with CUBE HA data GigabitEthernet 0/0/2
data GigabitEthernet 0/0/2
timers delay 30 reload 60 timers delay 30 reload 60
voice service voip Turn on CUBE voice service voip

mode border-element Redundancy mode border-element
allow-connections sip to sip allow-connections sip to sip
redundancy-group 1 redundancy-group 1
CUBE Configuration on ASR1K/ISR-4K/vCUBE Box-to-Box
Redundancy
CUBE 1 CUBE 2
track 1 interface GigabitEthernet 0/0/0 track 1 interface GigabitEthernet 0/0/0
line-protocol line-protocol
track 2 interface GigabitEthernet 0/0/1 track 2 interface GigabitEthernet 0/0/1
line-protocol Track interfaces line-protocol
to trigger
redundancy switchover redundancy
application redundancy application redundancy
group 1 group 1
track 1 shutdown track 1 shutdown
track 2 shutdown track 2 shutdown
Redundancy
CUBE 1 CUBE 2
interface GigabitEthernet0/0/0 Inside interfaces : interface GigabitEthernet0/0/0
description “Enterprise LAN” description “Enterprise LAN”
Redundancy Interface ip vrf forwarding LAN-VRF
ip vrf forwarding LAN-VRF
Identifier 1 ip address 10.10.1.2 255.255.255.0
ip address 10.10.1.1 255.255.255.0
redundancy rii 1 VRF ID : LAN-VRF (if redundancy rii 1
redundancy group 1 ip 10.10.1.3 exclusive applicable) redundancy group 1 ip 10.10.1.3 exclusive
interface GigabitEthernet0/0/1 interface GigabitEthernet0/0/1

Outside interfaces: description “Enterprise WAN”
description “Enterprise WAN”
ip vrf forwarding WAN-VRF Redundancy Interface ip vrf forwarding WAN-VRF
ip address 20.20.1.1 255.255.255.0 Identifier 2 ip address 20.20.1.2 255.255.255.0
redundancy rii 2 VRF ID : WAN-VRF (if redundancy rii 2
redundancy group 1 ip 20.20.1.3 exclusive applicable) redundancy group 1 ip 20.20.1.3 exclusive
Redundancy
CUBE 1 CUBE 2
interface GigabitEthernet 0/0/2 RG Control/Data interface : interface GigabitEthernet 0/0/2

ip address 3.3.1.1 255.255.255.0 CANNOT HAVE VRFs associated ip address 30.3.1.2 255.255.255.0
Redundancy
Configuration on Active and Standby
description to-SIP-SP Bind traffic destined to the outside (SP SIP
destination-pattern 9T trunk) to the outside Physical interface to
session protocol sipv2 make sure it uses the virtual IP address as
session target ipv4:y.y.y.y the source-IP for all calls
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
!
dial-peer voice 200 voip Bind traffic destined to the inside (CUCM or
description to-CUCM IP-PBX) to the inside Physical interface
destination-pattern 555….
session target ipv4:10.10.1.10
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
!
ip rtcp report interval 3000
! Configure media inactivity feature to
gateway clean up any calls that may not
media-inactivity-criteria all disconnect after a failover
timer receive-rtcp 5
timer receive-rtp 86400
Additional Supported options for CUBE HA .
M/
6.3 .1
CUBE-1 15
16.
3
PortChannel2
Gig0/0/0 XE
Gig0/0/2.200
Gig0/0/2.100 – ITSP 1
Gig0/0/1
CUBE
Gig0/0/3 Gig0/0/4 ITSP 1
PortChannel34
redundancy
redundancy
redundancy
vPC
WAN
rii 1
rii 2
rii 3
Switch D Switch E Switch A Switch B Switch C Edge
vPC
PortChannel34
CUCM ITSP 2
Gig0/0/3 Gig0/0/4
Gig0/0/1
PortChannel2
Gig0/0/2.100
Gig0/0/2.200 – ITSP 2
Gig0/0/0 CUBE
CUBE-2
• 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#CiscoLiveLA
for the LAN side in the above diagram withitsL2
© 2018 Cisco and/or and
affiliates. CEreserved.
All rights routerCisco
redundancy
Public
Additional Supported options for CUBE HA .
M/
6.3 .1
15 3
CUBE-1 16.
XE
CUBE
TenGig0/0/3 TenGig0/0/4
PortChannel34.30
vPC
redundancy redundancy
Switch A Switch B
rii 1 rii 2 Switch C
WAN Edge
PortChannel34.30
CUCM TenGig0/0/3 TenGig0/0/4
CUBE
CUBE-2
Agenda
CUBE + SRST Colocation
Cisco Unified Border Element – Router Integration
An Integrated Network Infrastructure Service
CUBE and SRST Cisco Unified Border Element TDM Gateway

§ Address Hiding
Prior to SRST 12.1 PSTN Backup
§ H.323 and SIP interworking
SCCP SRST on § DTMF interworking
ISR G2 w/CUBE § SIP security
Voice Policy
is supported § Transcoding CUBE
Note: An SBC appliance would
have only these features
SIP SRST on ISR
G2 w/CUBE is
not supported WAN & LAN IP Routing &
Physical MPLS
Any SRST on ISR Interfaces
4K with CUBE Unified CM
SRST
co-located was Conferencing and
Transcoding
not supported VXML FW, IPS,
QoS
.1
1 6.7
-XE 12.1
How Call Control works on an ISR platform I O S
SR
ST
HUSALI-SRST#show dial-peer voice summary

dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
911 voip up up 911$ 0 syst ipv4:5.5.5.5
20001 pots up up 2222$ 0 50/0/1
• SCCP CME/SRST creates virtual voice port (EFXS) for a registered SCCP Phone for
receiving and sending calls
• Each ephone-dn creates a virtual pots dial-peer beginning with 20XXX TAG
• SIP CME/SRST creates virtual voip dial-peers (voice register dn <tag>) beginning with
40XXX TAG
Branch CUBE Deployment with SRST Provisioned
Branch with Unified SRST Provisioned
on the same platform as CUBE
Unified CM LAN Dial-Peers WAN Dial-Peers
CUBE
Gig0/0
PSTN
Enterprise Gig0/1
Data IP WAN
Center RTP
SIP - Trunkside
SIP - Lineside
SIP Endpoints
Branch losing connectivity to Unified CM across the
Enterprise WAN
Branch with no WAN connectivity
Unified CM LAN Dial-Peers WAN Dial-Peers
CUBE
Gig0/0
PSTN
Enterprise Gig0/1
Data IP WAN
Center RTP
SIP - Trunkside
SIP - Lineside

Branch Operating in SRST mode with CUBE SIP
Trunk for PSTN calling
Branch in SRST mode
Unified CM LAN Dial-Peers (Busied WAN Dial-Peers (Unchanged)
Out)
SRST
Gig0/0
PSTN
Enterprise Gig0/1
Data IP WAN Voice Register

Center Pool / DNs
RTP
SIP - Trunkside
SIP - Lineside

.1
1 6.7
-XE 12.1
How Call Control works on an ISR platform I O S
SR
ST
HUSALI-SRST#show dial-peer voice summary

dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
20001 pots up up 2222$ 0 50/0/1
• SCCP CME/SRST creates virtual voice port (EFXS) for a registered SCCP
Phone for receiving and sending calls
• Each ephone-dn creates a virtual pots dial-peer beginning with 20XXX TAG
• SIP CME/SRST creates virtual voip dial-peers (voice register dn <tag>)

beginning with 40XXX TAG
.1
6.7
CUBE – SRST Co-Location 1
-XE 12.1
IOS RST
S
• Co-location of Unified SIP SRST and Unified Border Element is supported only on
ISR4000 series platforms.
• Validated for all Cisco SIP IP Phones including 69XX, 78/7900, 8800 and 9900 Series.
Not validated for SCCP endpoints. Non-Secure Jabber also supported.
• Not supported with E-SRST and Secure-SRST.
• During WAN outage, phones registered to Unified SIP SRST can place or receive PSTN
calls through CUBE’s ITSP SIP trunk.
• Dynamic dial-peers are created for phones falling back to Unified SRST; voice class
tenant configuration is leveraged to segregate trunk-side and line-side features than
global config (voice service voip, sip-ua)
CUBE – SRST Co-Location Configuration IOS-XE
16.7.1
Recommendations SRST 1
2.1
• Move ITSP SIP trunk specific voice service voip and sip-ua configurations
under voice class tenant and apply that tenant to WAN Dial-peers.
• This avoids configuration conflict between ITSP specific SIP trunking
requirements and Enterprise LAN side dial-peers. When a tenant is
configured and applied under a dial-peer, the IOS-XE configurations are
applied in the following order of preference:
• Dial-peer configuration
• Tenant configuration
• Global configuration (voice service voip / sip-ua)
• Do not configure “incoming called-number .T” on an inbound LAN dial-peer
CUBE – SRST Co-Location Configuration IOS-XE
16.7.1
Recommendations SRST 1
2.1
• The virtual dial-peers (40XXX TAG) created after the phones (registered to Unified
Communications Manager) fall back on Unified SRST are dynamic dial-peers.
Configurations under voice service voip and sip-ua are inherited by these dynamic dial-
peers
• Dial-peer groups feature should only be used on an Inbound LAN Dial-peer. Do not
configure dial-peer groups on the Inbound WAN dial-peers
• Configure SIP Options Request Keepalives to monitor reachability towards Unified

Communications Manager on Outbound LAN Dial-peers
• If dial-peers are using voice class codec, configure the same voice class codec under
voice register pool
CUBE – SRST Co-Location Feature Set 6.7
.1
1
XE
• SRST Line Side Features • SIP Trunk Features
§ Incoming & Outgoing Basic § SIP-UA

Call § Outbound Proxy
§ Hold-Resume § TLS/SRTP - UDP/ RTP Interworking
§ Call Forward § IP Trust List
§ Call Transfer § Voice Class Tenant
§ Conference (Built-in Bridge) § RTP-NTE DTMF
§ Hunt Groups § SIP Normalization
§ Bind Global /Dial-peer
§ PAI
Agenda
Key Takeaways
• CUBE + ZBFW is now supported on ISR 4K and recommended release is 16.6.3 onwards
• TLS 1.2, wide variety of ciphers
• MiFiD Compliance support from Nov 2018

• Newer platforms support such as ISR1100 series from April 2019
• Enterprise SBC (Cisco Unified Border Element - CUBE) is an essential component of a UC solution
providing;
• Security, Session Management, Interworking, Demarcation
• Over 36,000 Enterprise customers all over the Globe
• Proven interoperability with 3rd party PBX vendors and different service providers around the world (more
than 165 countries)
• Now is the time to deploy SIP Trunking in either a Centralized or a Distributed solution to save money,
simplify your topology and setup your infrastructure for future services
• Complete feature Presentations, Lab Guide, Free Hands-on Lab access & Application Notes
https://cisco.box.com/cube
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#CiscoLiveLA Presentation ID © 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 132
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SIP Trunk Design with Cisco

Unified Border Element

Webex Teams will be moderated cs.co/ciscolivebot#BRKCOL-2125

3900 Series ISR-G2 (3925, 3945)

4 <50 500-600 900-1000 2000-2500 4000 4500-6000 7000-10,000 12K-14K 14-16K

Active Concurrent Voice Calls Capacity

Proven Interoperability and Interworking with

Platform CUBE SIP-SIP IPT Sessions (Audio)

CPS1 Total # of SW CUBE + CUBE + Xcoding2

NOTE : Subject to Change

§ Actively involved in the call treatment, signalling

§ Media is handled in two different modes: CUBE

dial-peer voice 1 voip dial-peer voice 2 voip

VRF1 – 10.10.10.10 CUBE 20.20.20.20 – VRF2

180 RINGING 180 RINGING

• CSR (Cloud Services Router) 1000v runs on a Hypervisor – IOS

RP (control plane) ESP (data plane) FFP code

Chassis Mgr. QFP Client Chassis Mgr.

CUBE signaling CUBE media processing

CSR 1000v (virtual IOS-XE)

Hypervisor vSwitch NIC

X86 Multi-Core CPU Memory Banks Hardware GE … GE

2. Configure any other global settings to meet SP’s requirements

• Dial-Peer – “static routing” table mapping phone numbers to interfaces or IP addresses

voice class sip-profiles 100

New rule tagging mechanism is being introduced XE

CUBE(config)#voice class sip-profiles 1 For inserting a rule

voice class sip-profiles 1

rule before 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”

request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”

rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”

rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”

request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”

CUBE(config-class)#request INVITE sip-header WORD ?

Inbound LAN Dial-Peer Outbound Calls Outbound WAN Dial-Peer

voice class uri 1001 sip

A dial-peer voice 1 voip Inbound Calls

1 URI of incoming Address INVITE sip:654321@10.2.1.1 SIP/2.0

Outbound LAN Dial-Peer Inbound Calls Inbound WAN Dial-

200 OK • The dial-peer is “busyout” if it does

• Each dial-peer that has OPTIONS message configured

voice class server-group 1 dial-peer voice 100 voip

* DNS target not supported in server group

voice class e164-pattern-map 300 Provides the ability to combine multiple

G729 Sites A SIP Trunk SP SIP Trunk

Site A voice class e164-pattern-map 400 ! This is an example of the contents of

• Allows grouping of outbound dial-peers based on an incoming dial-peer, reducing

voice class dpg 10000 dial-peer voice 1001 voip

• Most commonly used for DTMF interworking, IPv4-IPv6

CUCM CPU Exception : Can

• Transcoding (12.4.20T) • Transcoding: G.711, G.723.1, G.726, G.728,

Histogram for Call rate 1122222357676678753222211111122247545789774322213311112245654598843333222

dial-peer (default) dial-peer sort dial-peer sort descending

Total Number of Active Calls XE

Router# show call active total-calls

Forked call (media forking) 3 2

May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/verbose/32/sipSPI_ipip_store_channel_info: dtmf negotiation

• CUCM NBR (Network Based Recording)

additionally be used to media-recording 950 Needs to

1. Enable HTTP on IOS

2. Enable the API on IOS

3. Enable XMF service within the API

[1] – [3]: An external call is answered by user with IP phone