Designing & Deploying UC Networks

Designing & Deploying UC networks with Cisco
Session Management Edition

BRKUCC-2931
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BRKUCC-2931 © 2012 Cisco and/or its affiliates. All rights reserved. Cisco Public 2
Session Objectives
 At the end of this session you should :
 Have a good understanding of how to design SME deployments
 Understand the benefits of SME designs
 Understand how to size an SME cluster
 SME DEPLOYMENT GUIDE

The content of this presentation is based on the SME Deployment
Guide which can be found here :
http://www.cisco.com/en/US/products/ps10661/products_implementati
on_design_guides_list.html
Agenda
 What is a Session Management Edition (SME) cluster ?
 Trunk Aggregation and Trunk Design recommendations
 Call Re-routing in SME deployments
 SME Deployments – Secure Trunks – Encrypted Signaling and Media
 Dial Plan Aggregation and Number Transformation
 Call Admission Control within SME based UC networks
 Signaling Delay considerations - Regional SME clusters
 QSIG in SME Deployments
 Mobility and SME
 UC Applications and SME
 SME Deployments with Unified Contact Centre Enterprise
 Sizing the SME cluster
 Summary
What is a Session Management Edition cluster ?
CUBE
CUBE
CUBE
H323 Trunk
MGCP Trunk
SIP Trunk
A CUCM cluster and SME cluster use exactly the same software
A CUCM cluster is typically used to register 10,000s of Phones
An SME cluster is typically used as a platform for Trunk and Dial Plan aggregation
Both CUCM and SME support Voice, Video and Encrypted calls
Support for SME deployments was introduced with UC version 7.1(2)
UC version 8.5 introduces a number of features that enhance SME functionality :
Improved SIP Trunk and H323 Inter Cluster Trunk functionality
Improved through-cluster routing
SIP Normalization and Transparency scripts

Why deploy Session Management Edition ?
An SME cluster at the core of your network allows you to flexibly manage your UC
system as it grows and changes…….
PSTN
Voicemail Conferencing
CUBE CUBE
System System
H323
Unified CM MGCP
Session SIP
Management
Edition Cluster
CUCM PBXs/ CUCM CUCM PBXs with PBXs with CUCM/PBX

Clusters with Clusters with Clusters with MGCP QSIG MGCP Q931 with SIP QSIG
H323 QSIG SIP Trunks H323 Trunks to Trunks to SME Gateway Trunks to SME
Trunks to SME to SME SME Trunks
Leaf Unified CM Clusters/ Leaf UC Systems
An SME cluster can interconnect 1000’s of UC systems using SIP, H323, or

MGCP Trunks SME allows you to reduce UC system complexity by centralizing
your dial plan and call routing rules in the SME cluster which in turn allows you to
simplify the dial plan and management of the connected UC systems
Reasons for deploying an SME cluster
PSTN
Voicemail CUBE CUBE Conferencing

System System
H323
Unified CM MGCP
Session SIP
Management
Edition
Cluster
PBXs/ CUCM CUCM

CUCM Clusters with PBXs with PBXs with MGCP CUCM/PBXs with
Clusters with Clusters with MGCPTrunks to Q931 Gateway SIP QSIG Trunks
SIP Trunks to H323 Annex M1
H323 QSIG Trunks to SME SME Trunks to SME
Trunks to SME SME
A centralized Dial Plan in SME allows you to :

Globalize and Normalize the called and calling numbers used by all Leaf systems
Manage overlapping number ranges in Leaf systems
Configure “find me” call routing using Route List and Route Groups
Re-route calls via the PSTN when the device cannot be reached via and IP path
SME Mobility features allow you to offer Single Number Reach functionality to
devices on 3rd Party UC systems
SME Normalization scripts allow you to modify any inbound or outbound SIP
message and SDP body content – simplifying interoperability with 3rd party UC
systems
SME cluster - PBX aggregation
SME migration to a CUCM cluster PSTN
CUBE CUBE
PSTN PSTN PSTN PSTN PSTN PSTN
Phase 1 – PBX Phase 2 – Centralized IP

Aggregation PSTN
PSTN
PSTN
CUBE CUBE
CUBE CUBE
Phase 3 – PBX Phase 4 – CUCM cluster

decommissioning
Trunk Aggregation and Trunk
Design recommendations for SME
deployments
Trunk Aggregation and Trunk Design
recommendations
 Summary deployment recommendations for SME Trunks

 Inter Cluster Trunk feature comparison
 New Trunk features that simplify SME designs
 Load Balancing, availability and redundancy
 SME Trunks to IOS gateways and 3rd Party UC systems
 CUCM Leaf cluster Trunks – UC version design considerations
 SME Clustering Over the WAN – design considerations
Summary design and deployment
recommendations for SME Trunks
 UC version 8.5+ is recommended for SME clusters as it introduces a range

new Trunking features that simplify SME deployments
 UC version 8.5+ is preferred for CUCM Leaf clusters as this allows them to
use the new Trunking features in this release, but earlier versions are also
supported
 With Leaf CUCM cluster UC version 8.5+ SIP ICTs and Trunks are preferred
 With Leaf CUCM cluster UC versions prior to 8.5 – H323 ICTs are preferred
 Although MGCP Trunks to gateways offer benefits in terms of ease of dial

plan configuration – they do not (cannot) benefit from the SIP and H323
Trunk features introduced with UC version 8.5
Unified CM 8.5 - Inter Cluster Trunks
SIP Trunks vs H.323 Trunks – Feature Comparison
H.323 (Q.SIG) SIP (QSIG)
Support for “+” character
Signaling Authentication and Encryption TLS
Media Encryption
“Run On All Nodes” feature
“Up to 16 destination addresses” feature
Calling Line ID / Name – Presentation / Restriction
Connected Line ID / Name – Presentation / Restriction
OPTIONS Ping
iLBC, AAC, ISAC and G.Clear Support
G.711, G.722, G.723, G.729 Support
SIP Subscribe / Notify, Publish – Presence
QSIG Path Replacement
QSIG Call Completion – No Reply / Busy Subscriber
Topology Aware - RSVP Based Call Admission Control
Message Waiting Indicator (On /Off)
Video / T.38 Fax support
BRKUCC-2931 Legend: Yes Limited support

© 2011 Cisco and/or its affiliates. All rights reserved. Cisco Public
No 12
New Trunk features that simplify SME designs
 UC version 8.5 introduces the following Trunk features :
 H323 Inter Cluster Trunks

o Run On All Unified CM Nodes
o Up to 16 Destination IP addresses
 SIP Trunks and SIP Inter Cluster Trunks

o Up to 16 Destination IP addresses
o SIP Options Ping
o SIP Normalization Scripts
 Route Lists
In conjunction with the “Route Local” feature – these new features

greatly simplify Leaf CUCM cluster and SME cluster Trunk design
and deployment
 Run On All Unified CM Nodes

 Up to 16 Destination IP addresses
 These features :
o Reduce the number of configured Trunks
o Simplify call distribution
o Simplify call routing through Leaf and SME clusters by taking
advantage of the “Route Local” rule…..
SIP/H323 SIP/H323
ICT Trunk ICT Trunk
SIP Trunk A
SIP Trunk A CUB CUB
E E
SIP Trunk B
CUB CUB
E E
CUCM SME CUCM SME
 Outbound SIP Trunks, H323 Inter Cluster Trunks and Route Lists can
take advantage of the Route Local Rule by using the “Run On All
Unified CM Nodes” feature
 The Route Local Rule
 If the CUCM node that the inbound call arrives on – also has an
instance of the selected outbound trunk for that call – then use this
node to onward route the call
 The Route Local rule reduces (and can eliminate) call set up traffic
between CUCM nodes with a cluster
SIP/H323 SIP/H323
ICT Trunk ICT Trunk SIP Trunk A
SIP Trunk
CUB CUB
E E
SIP Trunk B
CUB CUB
E E
Route List
UC 8.5 SIP Trunk Feature - OPTIONS Ping
SIP ICT Trunk
SIP Trunk
CUBE
CUBE
OPTIONS Ping is activated on a per SIP Trunk basis

Each node running the SIP Trunk daemon in the originating cluster uses
OPTIONS Ping to determine the availability of each defined destination IP
address
- CUCM will not attempt to establish a call to an unavailable remote peer

- SIP Trunk - “In Service” whilst one remote peer is reachable
- SIP Trunk - “Out Of Service” state when all remote peers are unreachable
- CUCM 8.5 SIP Trunks – Dynamic reachability detection

- Pre CUCM 8.5 Trunks - Per call time out
Unified CM Trunks – Load Balancing,
Availability & Redundancy – Summary UC 8.5
Features
HA Features affecting Calls originating from HA Features for Trunk

a CUCM cluster destinations
Calls over single Trunks Calls over multiple Trunks
Route Lists and Route Up to 16 Destination IP

Groups Addresses
H323 ICTs and SIP Trunks
Call Manager Groups Call Manager Groups H323 Gatekeeper

DNS SRV – SIP Trunks
Run on All Nodes Run on All Nodes SIP OPTIONS Ping

H323 ICTs and SIP Trunks H323 ICTs, SIP Trunks and SIP Trunks only
Route Groups
Route Local Route Local
UC 8.5 SIP Trunks – Normalization Scripts
SIP/SCCP SIP Trunk
Normalization allows incoming and
outgoing SIP messages to be modified
Script on their way through a CUCM SIP Trunk.
SIP Trunk SIP Trunk IP PSTN The Normalization feature is designed to

improve interoperability between CUCM
Script
SIP Trunks and SIP based 3rd Party SIP
H323 Trunk SIP Trunk
PBXs, Applications & IP PSTN services.
Script Normalization is independent of what the

SIP Trunk connects to on the other side
MGCP Trunk SIP Trunk IP PSTN of CUCM. e.g.
Normalization Script • SIP SIP Trunk calls

Remove “display name” from SIP Header • Skinny SIP Trunk calls
function fixInboundPai() • H.323 SIP Trunk calls
local pai = Sip.getHeader("P-Asserted-Identity") • MGCP SIP Trunk calls
local displayname = getDisplayName(pai)
local uri = getUri(pai)
local number = getUserPart(uri) Normalization uses a scripting
if displayname == number environment to allow customers to
then modify SIP messages and SDP content
Sip.modifyHeader("P-Asserted-Identity", uri) on a per trunk basis.
Scripting BRKUCC-2931
Guide at : http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/sip_tn/8_5_1/sip_t_n.html
© 2011 Cisco and/or its affiliates. All rights reserved. Cisco Public 18
UC 8.5 SIP Trunks – Transparency Scripts
SIP Trunk SIP Trunk
Transparency allows CUCM to pass
headers, parameters, and SDP content
Script from one SIP call leg to the other.
SIP Trunk SIP Trunk The Transparency feature is designed

to improve the operation of and
Script
interoperability between 3rd Party SIP
SIP Trunk SIP Trunk
PBXs and Applications connected via
CUCM/SME.
Script
Transparent pass through is only
Transparency Script applicable when the call through
Sip.allowHeader(“A-Callid”)
Sip.allowHeader(“A-ConversationId”)
CUCM is from SIP Trunk to SIP Trunk.
function A.inbound_INVITE
Sip.passThroughHeader(“A-Callid”) • SIP Trunk SIP Trunk calls
Sip.passThroughHeader(“A-ConversationId”)
Sip.passThroughHeaderValue("Supported", "x-
Transparency uses the same scripting
nortel-sipvc")
Sip.passThroughUriParameters("From", environment as Normalization to allow
"uriparm1") customers to pass SIP messages
Sip.passThroughHeaderParameters("From", through CUCM. Transparency and
"hparm1", "hparm2") Normalization features can be combined.
end
Scripting BRKUCC-2931
Guide at : http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/sip_tn/8_5_1/sip_t_n.html
SIP Trunks vs H323 and MGCP Trunks to gateways
 SIP Trunks support the “Run On All Unified CM Nodes” and “Up to 16
destination IP addresses” features
 H323 and MGCP Trunks to gateways and 3rd party UC systems support
standard Call Manager Groups and a single destination IP address
 Using standard Call Manager Groups (rather than Run on All Nodes)
increases call set up traffic between nodes within a cluster
 Multiple Trunks may be required with the single destination IP address
limitation
 Note – MGCP Trunks are only active on one node in the Call Manager
Group (as the signaling channel is back hauled to CUCM)
H323 SIP
ICT Trunk ICT Trunk MGCP Trunk A
H323 Trunk A
H323 Trunk B MGCP Trunk B
Route List Route List

BRKUCC-2931 Selected
© 2011 Cisco and/or outbound
its affiliates. All Trunk
rights reserved. Cisco Public Selected outbound Trunk 20
IOS Gateway Trunks
SIP Trunks , H.323 Gateways, MGCP Gateways – Feature Comparison
H.323 SIP MGCP

Centralized Provisioning
QSIG Tunneling
Centralized CDR (DS0 Granularity in Unified CM CDR)
MLPP (Preemption)
Hook-flash Transfer with Unified CM

ISDN Overlap Sending No GK
NFAS
SRTP (Unified CM to GW)
CUCM 8.5 “Run On All Nodes” feature 3 Active Nodes in 1 active Node
a CMG in a GMG
CUCM 8.5 “Up to 16 destination addresses” feature
Mobility Manager VXML-Based Voice Profile Mgmt
OPTIONS Ping
TCL/VXML Apps (e.g. for CVP Integration)
Voice & Data Integrated Access
Fractional PRI Workaround
TDM Variations: A-DID, E&M, PRI NFAS, CAMA, T1 FGD

ISDN Video Switching on GW
Set numbering Plan Type of Outgoing Calls
G.Clear (Clear Channel Data) Support
H.320 Video
Legend:
BRKUCC-2931 Yes
© 2011 Cisco and/orLimited
its affiliates. Allsupport
rights reserved. No
Cisco Public CMG – Call Manager Group 21
SME Trunk Design Considerations
Multi-Cluster Designs – with CUCM 8.5 Leaf Clusters
IP
SIP Delayed Offer SIP PSTN
SIP Early Offer CUBE SIP DO to EO

CUBE CUBE
CUCM 8.5
CUCM 8.5
CUCM 8.5 SME
SIP ICT Trunks - Voice, Video and Encryption supported

OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, Multiple Destination Addresses,
QSIG over SIP. (Call Back & Path Replacement)
SIP IP PSTN Trunks – Typically Voice only – Early Offer usually required by SP
OPTIONS Ping, SIP Delayed Offer, If EO required - CUBE provides SIP DO to EO, Run
on All Nodes, Multiple Destination Addresses. (EO should be sent by the SP)
SIP to 3rd Party UC Systems – Typically Voice – Video & Encryption also supported
OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, If required - QSIG over SIP, If
end device is capable it should send Early Offer.
SME Trunk Design Considerations
Multi-Cluster Designs – with Pre 8.5 CUCM Leaf Clusters
SIP Delayed Offer SIP IP
PSTN
SIP Early Offer H323
CUBE SIP DO to EO
H323 Slow Start CUBE CUBE
CUCM 8.0
CUCM 7.X
CUCM 8.5 SME
Pre 8.5 Clusters - H323 ICT Trunks - Voice, Video and Encryption supported
H323 Slow Start, Call Manager Groups, 3 Destination Addresses per Trunk, QSIG
over H323.
OPTIONS Ping, SIP Delayed Offer, If EO required - CUBE provides SIP DO to EO,
Run on All Nodes, Multiple Destination Addresses. (EO should be sent by the SP)
SIP to 3rd Party UC Systems – Typically Voice – Video & Encryption also supported
OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, If required – QSIG over SIP, If
end device is capable it should send Early Offer.
SIP Trunk Design Considerations
Multi-Cluster Designs – Mixed Leaf Cluster CUCM Versions
SIP Delayed Offer SIP IP
PSTN
SIP Early Offer H323
CUBE SIP DO to EO
H323 Slow Start CUBE CUBE
CUCM 8.0
CUCM 8.5
CUCM 8.5 SME
Pre 8.5 Leaf Clusters – H323 ICT Trunks – Voice, Video and Encryption supported
H323 Slow Start, Call Manager Groups, 3 Destination Addresses per Trunk, QSIG over
H323.
8.5 Leaf Clusters – SIP ICT Trunks – Voice, Video and Encryption supported
SIP Delayed Offer, OPTIONS Ping, Run on All Nodes, Multiple Destination Addresses,
QSIG over SIP.
OPTIONS Ping, SIP Delayed Offer, If EO required – CUBE provides SIP DO to EO, Run
on All Nodes, Multiple Destination Addresses. (EO should be sent by the SP)
Clustering Over the WAN – Calls from Leaf Clusters to SME
IP
PSTN COW Bandwidth requirements as
SIP Delayed Offer CUBE CUBE
per the UC SRND - 80mS RTT
West Coast DC
San Jose New York
Primary Secondary
SME 8.5
CUCM 8.5 CUCM 8.5
San Francisco
East Coast DC SIP
CUBE SIP DO to EO
CUBE CUBE
IP
CUCM 8.5 PSTN
In each Leaf Cluster :

• Create and prioritize multiple SIP Trunks in Route Lists to distribute calls to
• each group of SME nodes in each Data Centre, Run Route Lists on all Nodes
• Enable “Run on all Nodes” on each Leaf Cluster SIP Trunk.
• Define destination IP addresses per Trunk for geographic call distribution
• Use SIP Delayed Offer (DO) on all Trunks (reduces MTP usage)
• For IP PSTNs - If Early Offer (EO) is required – Use CUBE SIP DO to EO feature
Clustering Over the WAN – Calls from SME Cluster to Leafs
IP
PSTN COW Bandwidth requirements as
SIP Delayed Offer CUBE CUBE
per the UC SRND - 80mS RTT
West Coast DC
San Jose New York
West West
SME 8.5
CUCM 8.5 CUCM 8.5
San Francisco
East Coast DC SIP
CUBE SIP DO to EO
CUBE CUBE
CUCM 8.5 IP
PSTN
In the SME Cluster :

• Use standard Call Manager Groups for each SIP Trunk – Define destination IP
• addresses for every call processing node in the Leaf cluster
• For routes to each Leaf cluster - Add SIP Trunks to a Route Group and Route
• List. Run Route Lists on all Nodes. Use Local Route Groups to route outbound
• calls over Trunks in the same data centre as the inbound Trunk call
• Use SIP Delayed Offer (DO) on all Trunks (reduces MTP usage)
• For IP PSTNs - If Early Offer (EO) is required – Use CUBE SIP DO to EO feature
Call Re-Routing in SME deployments
Call Re-routing in SME Deployments
Single outbound SME Trunk
No Circuit Available
SME
CUBE
IP
PSTN
IP
PSTN
CUBE
SME x
Call Admission Control
Out Of Bandwidth
Call Progress failure messages (e.g. No circuit, Number Unavailable,

No bandwidth etc) are passed back to the originating Leaf cluster –
which can then re-route the call
Call Re-routing in SME Deployments
Multiple outbound SME Trunks in a Route List
IP
PSTN
CUBE CUBE
No Circuit Available
SME
IP
PSTN
CUBE CUBE
SME
x
Call Admission Control
Out Of Bandwidth
Call Progress failure messages (e.g. No circuit, Number
Unavailable, No bandwidth etc ) are passed back to the SME
cluster – which can then re-route the call
Secure Trunks
Encrypted Media and Signaling
SME Deployments – Secure Trunks
– Encrypted Signaling and Media
 Secure Signaling and encrypted Media can be configured

for SIP and H323 Trunks
 SIP Trunk security using TLS is preferred over H323
- H323 uses IPSEC and requires IPSEC Tunnels to be
set up in the network rather than on CUCM/SME
nodes
 Using central Certificate Authority with SIP Trunks reduces
configuration overhead and simplifies certificate
management
 Media Encryption (SRTP) can be set up without secure
signaling but in this case security keys are sent in the
clear
Secure SIP Trunks using TLS and an external CA
Certificate Authority (CA)
CA Certificate
CA Signed CUCM
Certificate
Leaf Cluster A
SIP Inter Cluster Trunk using

TLS for signaling encryption
Session
Encrypted Management
Media Edition
Cluster
Leaf Cluster B
Secure H323 Trunks using router based IPSec Tunnels
Leaf Cluster A
H323 Inter Cluster Trunk using router
based IPSEC for signaling encryption
IPSEC Tunnel
Session
Management
Encrypted Edition
Media Cluster
Leaf Cluster B
Dial Plan Aggregation and Number
Transformation
Dial Plan Aggregation and Number
Transformation
 Basic Principles
 Number Transformation options
 Dial Plan and dialing habit variations
 Number Transformation capabilities by Trunk protocol type
SME Dial Plan Aggregation – Basic Principles
In the SME cluster a unique globalized dial plan should be

implemented – Inbound number globalization and outbound
number localization can be implemented in the SME cluster
SME Cluster
Leaf Cluster Leaf UC system
In Leaf clusters/ Leaf UC systems a localized dial plan

may be implemented and users may have local
dialing habits
SME - Number Normalization Tools
Inbound number globalization and outbound number
localization can be implemented in the SME cluster
SME Cluster
With Session Management Edition and CUCM, normalisation of calling and

called party information can be achieved by using a variety of tools :
• Transformation calling search spaces on trunk or device pool level

• Transformations on route patterns
• Transformations on translation patterns
• Transformations on route lists
This presentation focuses on number transformations at the Trunk level......

But first lets look at some variations on dial plan and dialling habit
implementations........
Internal Directory Numbers, +E.164 SME Dial Plan,
Abbreviated On Net Dialing
Calling Number +44-404-444-4000 Calling Number +44-404-444-4000
Called Number +55-505-555-5000 Called Number +55-505-555-5000
Cluster 1 DNs Cluster 2 DNs

444-4XXX SME Cluster 555-5XXX
+ E.164 Dial Plan

Called Number 8-555-5000 Called Number 555-5000
Caller (444-4000) dials 8-555-5000
Leaf Cluster 1 Leaf Cluster 2

Internal Directory Numbers (XXX-YYYY) Internal Directory Numbers (XXX-YYYY)
Abbreviated Dialing (8-XXX-YYYY) Abbreviated Dialing (8-XXX-YYYY)
Internal Directory Numbers, +E.164 SME Dial Plan,
Abbreviated On Net Dialing
All number transformations performed on SME cluster
Calling Number +44-404-444-4000 Calling Number 8-444-4000
Called Number +55-505-555-5000 Called Number 555-5000

444-4XXX SME Cluster 555-5XXX
+ E.164 Dial Plan
Calling Number 444-4000 Calling Number 8-444-4000

Called Number 8-555-5000 Called Number 555-5000
Caller (444-4000) dials 8-555-5000

Internal Directory Numbers (XXX-YYYY) Internal Directory Numbers (XXX-YYYY)
Abbreviated Dialing (8-XXX-YYYY)
BRKUCC-2931 © 2011 Cisco and/or its affiliates. All rights reserved. Cisco Public Abbreviated Dialing (8-XXX-YYYY) 39
+E.164 Directory Numbers, +E.164 SME
Dial Plan, E.164 Dialing

+44-404-444-4XXX SME Cluster +55-505-555-5XXX
+ E.164 Dial Plan

Caller (+44-404-444-1000) dials 55-505-555-5000

+ E.164 Directory Numbers + E.164 Directory Numbers
E.164 Dialing E.164 Dialing
SIP Trunks – Inbound and Outbound
Digit manipulation capabilities
Inbound SIP Trunk Outbound SIP Trunk
SME /CUCM
Cluster
Inbound Calling Search Space Outbound Called Party Transformation

AAR Calling Search Space CSS
Prefix DN – Prefix Digits to Incoming Called Outbound Calling Party Transformation
Number CSS
Incoming Calling Number – Prefix, Strip Caller ID DN – Overwrite/Prefix Digits to
Transform Digits Outbound Calling Number
+ char – carried in SIP + char – carried in SIP
+ char – carried in called and calling + char – carried in called and calling number
number QSIG IEs QSIG IEs
H323 Trunks and Gateways– Inbound and
Outbound Digit manipulation capabilities
Inbound H323 Trunk Outbound H323 Trunk
SME /CUCM
Cluster
Inbound Calling Search Space (CSS)

Outbound Called Party Transformation
AAR Calling Search Space
CSS
Prefix DN – Prefix Digits to Incoming
Outbound Calling Party Transformation
Called Number
CSS
Incoming Calling Number – Prefix, Strip
Caller ID DN – Overwrite/Prefix Digits to
Transform Digits
Outbound Calling Number
Incoming Called Number – Prefix, Strip
+ char – not carried in H323
Transform Digits
+ char – not carried in QSIG
+ char – not carried in H323
MGCP Trunks – Inbound and Outbound
Digit manipulation capabilities
Inbound MGCP Trunk Outbound MGCP Trunk
SME /CUCM
Cluster
Outbound Called Party Transformation

Inbound Calling Search Space
CSS
AAR Calling Search Space
Outbound Calling Party Transformation
Prefix DN – Prefix Digits to Incoming
CSS
Called Number
Caller ID DN – Overwrite/Prefix Digits to
Incoming Calling Number – Prefix, Strip
Outbound Calling Number
Transform Digits
+ char – not carried in Q.931
+ char – not carried in Q.931
Call Admission Control (CAC) in
SME based deployments
Call Admission Control within SME
based UC networks
 CAC Options :
 Locations based CAC
 RSVP based CAC
 Combinations of RSVP and Locations based CAC
 Local RSVP can be deployed in the SME cluster only

- This has advantages in deployments where partially meshed
international WAN circuits are used to interconnect SME and Leaf
clusters, or where sites are dual homed to the WAN
- RSVP can be used to accurately and efficiently use available WAN
bandwidth
Call Admission Control within SME based
UC networks
Both Locations based Call Admission Control (CAC) and RVSP
based CAC can be used in Leaf clusters and the SME cluster…..
 Locations based CAC for campus clusters provided by SME

 Locations CAC for Leaf clusters and SME using “hub none” locations
 Locations CAC for Leaf clusters and SME using “phantom” locations
 Local RSVP in the SME cluster

 Local RSVP in Leaf and SME clusters
 End to End RSVP in Leaf and SME clusters
 Mixed End to End RSVP in Leaf and SME clusters and Locations
based CAC in other Leaf clusters
SME cluster provides Locations based CAC
for campus clusters
SME ≥ 8.5
Leaf 1 Leaf 3
SIP Trunk
Registered Endpoint
Leaf 2
Location
Association
LOCATION 1 LOCATION 2
Associated to Trunks for Leaf 1 and Leaf 2 Associated to Leaf 3 Trunk (Leaf 3 is a single
(both Leaf Clusters have endpoints located in site cluster deployment)
the same physical site)
Multi Branch Leaf clusters provide Locations CAC – ICTs in
Hub_None Hub_None
Location
ICT-CL1 ICT-CL2 Hub-None
Hub_None SME Hub_None

ICT-SME ICT-SME
Central Site Central Site

Cluster 1 Cluster 2
Cluster 1 Locations CAC WAN Cluster 2 Locations

80k Deduction for CAC
Location B 80k Deduction for
Location F
SIP Trunk
Location Location F
B Call Signaling - 80 kbps
- 80 kbps Branch 1 Location Association Branch 2
BRKUCC-2931 Cluster 1 Cisco and/or its affiliates. All rights reserved.
© 2011 Cisco Public Cluster 2 48
Media
Multi Branch Leaf clusters - Locations CAC – Call Transfer
Hub_None Hub_None
ICT-CL1 ICT-CL2

ICT-SME ICT-SME

Cluster 1 Cluster 2
Cluster 1
Locations CAC Cluster 2
160k Deduction for
WAN Call Transferred to
Location B (2 Calls) Cluster 1 Location B
Transfer
SIP Trunk
Location Location F
B Call Signaling No WAN
- 160 kbps Branch 1 Location Association Branch 2 calls
Media
Multi Branch Leaf clusters provide Locations based CAC –
Phantom Locations allow the Phantom Phantom
ICTs in
originating location of each ICT-CL1 ICT-CL2
Location
call to be sent over ICTs
1 2
Phantom
Phantom SME Phantom
ICT-SME ICT-SME

Cluster 1 Cluster 2
Cluster 1 Locations CAC WAN Cluster 2 Locations

80k Deduction for CAC
Location B 80k Deduction for
Location F
SIP Trunk
Location Location F
B Call Signaling - 80 kbps
- 80 kbps Branch 1 Location Association Branch 2
Media
Multi Branch Leaf clusters - Locations CAC – Call Transfer
Phantom Locations allow the Hub_None Hub_None The originating location of
originating location of each ICT-CL1 ICT-CL2 the inbound transferred call
call to be sent over ICTs is returned to cluster 1
1 2

ICT-SME ICT-SME
Central Site 4 3 Central Site

Cluster 1 Cluster 2
Cluster 1
Locations CAC Cluster 2
0 kbps Deduction for WAN Call Transferred to
Location B Cluster 1 Location B
(0 WAN Calls)
Transfer
SIP Trunk
Location Location F
B Call Signaling No WAN
No WAN Branch 1 Location Association Branch 2 calls
calls
Media
SME cluster – Deploying Local RSVP
CAC for partially meshed WANs
10 WAN Link
UC Bandwidth IP PSTN
CUBE CUBE
100
10 10
1 1
10 10
For CUCM Leaf Clusters

Using Locations based Call Admission Control
What bandwidth value should be used for calls over SIP inter cluster Trunks into the WAN
?
For Voice Gateways
Using Call Counting based Call Admission Control
What value should be used for calls over SIP inter cluster Trunks into the WAN ?
Local RSVP in the SME cluster has advantages in deployments where

partially meshed international WAN circuits are used, or where sites are dual
homed to the WAN. RSVP does not need to be implemented in the leaf
clusters
SME cluster – Local RSVP CAC
Locations Based
CAC Local RSVP Local RSVP
SME ≥ 8.5
SCCP RSVP Agents registered to Leaf
and SME clusters may be co-
H323 Inter cluster
located on the same platform
Trunk
Leaf Cluster 1 Location Location Leaf Cluster 2

7.X or below Leaf 1 Leaf 2 7.X or below
Inter Location
RSVP Policy =
Mandatory
SME manages RSVP Agents

between Clusters
Local RSVP in the SME cluster has advantages in deployments where partially meshed
international WAN circuits are used, or where sites are dual homed to the WAN. RSVP does
not need to be implemented in the leaf clusters
SME cluster – Local RSVP CAC – Inter cluster Call Setup
Locations Based
CAC Local RSVP Local RSVP
SME ≥ 8.5
RSVP Agents registered to Leaf
Call Setup 1 3 Call Setup

7.X or below Leaf 1 2 Leaf 2 7.X or below
SME RSVP
Reservation
Leaf
Call Signaling RSVP
SCCP Resv.
H323 Inter cluster Trunk
RSVP
With Local RSVP in the SME cluster – Call set up proceeds on the outbound SME Trunk
once the RSVP Reservation has succeeded between the two SME controlled RSVP agents
SME cluster – End to End RSVP CAC with SIP
Preconditions
End-to-End RSVP with SIP Preconditions

SME ≥ 8.5
SCCP
SIP Inter cluster Trunk
RSVP

8.0 or above Leaf 1 Leaf 2 8.0 or above
SIP Preconditions passed over SIP Inter

cluster Trunks from Cluster 1 via SME to
Cluster 2
SME RSVP
Reservation
SME cluster – End-to-End RSVP with SIP
Preconditions extended into one Leaf cluster
Locations Based
CAC Local RSVP and End-to-End RSVP with SIP Preconditions
SME ≥ 8.5
SCCP
H323 Inter cluster
Trunk
SIP Inter cluster
Trunk

7.X or below Leaf 1 Leaf 2 8.0 or above
Inter Location
RSVP Policy =
Mandatory
SME RSVP
Reservation
SME cluster – Local RSVP CAC – Co-Locating RSVP
agents on the same router platform
sccp local GigabitEthernet0/0
sccp ccm 10.10.10.10 identifier 1 version
8.0
Local RSVP Local RSVP sccp ccm 10.10.20.20 identifier 2 version
7.0
SME ≥ 8.5 sccp ip precedence 3
sccp
RSVP Agents registered to Leaf
…
sccp ccm group 10
associate ccm 1 priority 1
associate profile 10 register
RSVP_Agent_SME
…
sccp ccm group 20
associate ccm 2 priority 1
associate profile 20 register
Location RSVP_Agent_Leaf2
Leaf 2 Leaf Cluster 2 …
dspfarm profile 10 mtp
codec g729r8
codec pass-through
rsvp
maximum sessions software 300
associate application SCCP
…
dspfarm profile 20 mtp
codec g729r8
SME and Leaf cluster RSVP codec pass-through
Agents co-located rsvp
maximum sessions software 300
on the same router platform associate application SCCP
Signaling Delay Considerations
Regional SME Clusters
Signaling Delay and SME based UC networks
SME
Recommendations for media delay are well defined

(ITU Recommendation G.114. < 150mS = accptable, 150 – 400mS = acceptable with
some impact on quality, > 400mS generally unacceptable)
Recommendations for signaling delay are not well defined

Primarily because the incurred delays are protocol dependent and the impact of long
delay generally affects call set up rather than overall voice quality
Impact of Signaling Delay on Call Set Up in SME
Messages exchanged before the
caller hears ringback tone networks
One way signaling Delay
INVITE INVITE
100 Trying 100 Trying
180 Ringing 180 Ringing
200 OK w/ SDP (Offer) 200 OK w/ SDP (Offer)

SME
ACK w/ SDP (Answer) ACK w/ SDP (Answer)
Two Way Media

Delay before the caller hears the Messages exchanged before called
called user after ringback stops user hears the caller after picking
up their handset
The diagram above shows an example of call set up delays and their impact on the
users’ experience. (Note – Phone to Call Agent signaling delay has been assumed to be
minimal)
Delays during call set up will vary based on the protocol(s) used, the trunk
configuration and call agent operation – making it difficult to calculate the time taken to
establish each stage of the call set up.
In most cases, signaling delays do not noticeably affect user experience. If signaling
delays are a concern, consider deploying regional SME clusters.
Regional SME clusters
IP IP
PST PST
N N
CUBE CUBE CUBE CUBE
SME SME
America Europe
SME
Asia Pac
CUBE
IP
PST
N
Provide a Regionalized SME cluster if there are multiple leaf UC systems in a

geographic region with significant amounts of intra region traffic
If regionalized SME clusters are not deployed, situate your SME cluster(s)
closest to those leaf UC systems that generate the most inter cluster traffic
QSIG and
SME deployments
QSIG and SME UC deployments
 With UC 8.5 - QSIG is supported over SIP, H323 and MGCP Trunks
 QSIG specific Features :
o Call Back on Busy
o Call Back on No Answer
o Path Replacement
o Calling Name
o Connected Name
 QSIG features such as Call Back and Path Replacement can use calling
and called numbers to determine whether the feature should be invoked.
Numbers carried in QSIG APDUs are not modified by CUCM number
transforms. In SME designs where number normalization is deployed –
Use the following CUCM/ SME service parameters :
o Call Back : Connection Retention (default setting)

o Path Replacement : PINX ID – Use a globally unique number for the PINX ID of
each cluster (The PINX ID is equivalent to any routable
number in the UC system and uniquely identifies a cluster)
Benefits of QSIG Path Replacement
in SME based UC deployments
IP QSIG over SIP
PSTN
QSIG over H323
CUBE CUBE
Media
1000
1001
9 10
4000
10 9
Phone 1000 calls Phone 4000
Signaling Call leg from 1XXX cluster via SME to 4XXX cluster
Call Admission Control deducts the bandwidth for a call from
1XXX to 4XXX
RTP media path direct between the two Phones over the WAN
Benefits of QSIG Path Replacement
in SME based UC deployments
IP QSIG over SIP
PSTN
QSIG over H323
CUBE CUBE
Media
1000
1001
8 10
4000
10 8
Phone 4000 Transfers the call to Phone 1001
New Signaling Call leg from 4XXX cluster via SME to 1XXX cluster
Call Admission Control deducts the bandwidth for a call from
4XXX to 1XXX
Resulting Media Path direct between Phone 1000 and Phone 1001
Benefits of QSIG Path Replacement in
SME based UC deployments
IP PSTN
SCCP/SIP
CUBE CUBE
Media
1000
1001
10 10
4000
10 10
QSIG Path Replacement for Tromboned calls
Clears down redundant signaling

Releases unused Call Admission Control bandwidth
For more info on QSIG and SME deployments see

The SME Deployment Guide :
http://www.cisco.com/en/US/products/ps10661/products_implementation_design_guides_list.html
and the CUCM Admin Guide :
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08procl.html#wp1139777
Cisco Mobility and SME
deployments
Mobility Features in SME based UC networks
SIP Delayed Offer SIP
H323
SIP Early Offer
H323 Slow Start
IP TDM
PSTN PSTN
CUBE CUBE
Unified Mobility Unified Mobility

User User
CUCM 8.5
CUCM 8.0
SME
PBX with SNR capability PBX using SME based

Mobility features
For SME based UC designs - Unified Mobility can be deployed as follows :
 Standard Leaf CUCM cluster mobility deployment – where all mobility features and PSTN
access are provided on the Leaf cluster
 Leaf cluster mobility with SME based PSTN access
 3rd Party PBX Mobility/ Single Number Reach support with PSTN access via SME
 Mobility features provided by SME for a connected 3rd party PBX

Leaf cluster mobility with SME based PSTN access
Note – SME PSTN
access can be TDM
or IP based
IP TDM
PST PSTN
N
CUBE CUBE

User User
CUCM 8.0 SME CUCM 8.5

 Supported Features
 Mobile Connect (aka SNR)
 Mobile Voice Access (MVA)
 Desk Phone pickup via hangup at mobile device and resume at desk phone.
 Remote Destination pickup using "Send Call to Mobile"
 Cisco Unified Mobile Communicator (CUMC)*
 Dual-mode Phones and Clients / Direct Connect (Cisco Jabber) Mobile Clients / BlackBerry MVS

 Unsupported Features - Features that use DTMF tones for feature invocation :
 Enterprise Feature Access (EFA)
 Desk Phone and Remote Destination pickup
 Mid-call Supplementary Services
* EOL 16th Dec 2011 – Migration Path to Cisco Jabber or MVS
Mobility features provided by SME for a 3rd party PBX
Note – SME PSTN
access can be TDM
or IP based
IP TDM
PST PSTN
N
CUBE CUBE

User User
PBX using SME

PBX with SNR based Mobility
capability SME features
Where a 3rd Party PBX does not natively support any mobility features, the SME cluster can
act as a mobility feature proxy by creating a Remote Destination Profile and two or more
Remote Destination numbers/ IDs per user. One Remote Destination will be the directory
number of the user’s 3rd Party PBX Phone and an additional Remote Destination number can
be the user’s mobile phone number.
Supported mobility features :
Mobile Connect (aka SNR)
Mobile Voice Access (MVA)
Unsupported Features - Features that use DTMF tones for feature invocation :
Enterprise Feature Access (EFA)
Desk Phone and Remote Destination pickup
Mid-call Supplementary
BRKUCC-2931 Services
UC Applications in SME
deployments
UC Applications in SME deployments
System System
Session H323
Management MGCP
Edition SIP
Cisco Cluster Cisco Unified
Cisco Unified Operator
Contact Emergency Presence
CER Console
Center Responder Server
SME supports the following centralized applications :

Unity, Unity Connection
Meeting Place, Meeting Place Express
SIP and H323 based Video Conferencing systems
3rd Party Voice Mail systems
Fax servers
Applications that track user or phone state must connect to the leaf cluster :
Unified Contact Centre, Unified Contact Centre Express
Cisco Unified Presence Server
Attendant Console
Cisco Emergency Responder
Centralized Unity VM in SME deployments
System System
Session H323
Management MGCP
Edition SIP
Cisco Cluster Cisco Unified
Cisco Unified Operator
Contact Emergency Presence
CER Console
Center Responder Server
On Trunks between leaf clusters and SME and on Trunk connections to your Voicemail application
ensure that the original called party/ redirecting number is sent with calls routed to voicemail.
For Non QSIG Trunks – Original Called Party /Redirecting number transport can be enabled by :
Enabling inbound and outbound Redirecting Diversion Header Delivery on SIP Trunks
Enabling inbound and outbound Redirecting Number IE Delivery on MGCP Gateways, H.323 Gateways
and H.323 Trunks
For QSIG enabled SIP, MGCP and H323 Trunks – the Original Called Party number is sent in QSIG
Diverting Leg Information APDUs. Note - The + character is not sent with the Diversion information sent
in QSIG APDUs.
For more information see the SME Deployment Guide :
http://www.cisco.com/en/US/products/ps10661/products_implementation_design_guides_list.html
SME Deployments with Extension Mobility Cross
Cluster
IP PSTN
CUBE CUBE
Unified CM EMCC Trunks

Session Inter Cluster
Management Trunks
Edition
Cluster
EMCC can be deployed as an overlay between leaf clusters in an SME based

UC network. Leaf cluster EMCC Trunks do not point to the SME cluster. In
each Leaf cluster, an EMCC Trunk(s) and its corresponding Remote Cluster
destination information is defined such that each EMCC Trunk can reach all
other EMCC enabled leaf clusters directly.
SME deployments with Cisco
Unified Contact Centre
SME Deployments with UCCE
 Unified CM SME can be deployed in Unified Contact Center
Enterprise (CCE) designs, where one or more leaf clusters are
contact center-enabled.
 There are specific requirements for the architecture used in these
scenarios :
- Unified CCE is only certified using SIP as the protocol
interconnecting the solution components.
- The preferred DTMF transport in Unified CCE is RFC2833 (“inband
DTMF”), since it is the only method supported for self-service
applications on Unified CVP
- Unified CCE and Unified CVP often use SIP headers to pass
information between systems. By default, SME does not
transparently relay some of these SIP headers and payloads from
one trunk to another. However, if required, a transparency script can
be applied to the SME trunk.
- See the SME Deployment Guide for more details
Back Office CUCM’s/PBX’s SUPPORTED DESIGN
Contact Centre
CUCM Cluster
Calls Transferred to
Unified CM Agents
Session
Management
Edition Cluster
Back-office calls JTAPI
IP PSTN
Contact
Centre calls
CUBE CUSP
Inbound calls to Contact Centre flow through CUSP and are Queued Calls
routed to a Queue (e.g. On VXML gateway) via CUSP
Calls transferred from the Queue to an Agent are routed

through SME and optionally CUSP
VXML
Calls to Back Office Users are routed via CUBE and SME Gateway
No Calls are transferred from Back Office to Contact Centre
Back Office CUCM’s/PBX’s NOT SUPPORTED
Contact Centre
CUCM Cluster
Unified CM
Session
Management
Edition Cluster
JTAPI
IP PSTN
CUBE
NOT SUPPORTED Queued Calls
Inbound calls to Contact Centre flow VXML

through SME and are routed to a Queue Gateway
(e.g. On VXML gateway)
IPv6 based
SME deployments
IPv6 based SME Deployments
PSTN
Voicemail CUBE CUBE

Conferencing
System System
Unified CM
Session SIP
Management
Edition Cluster
 Cisco UC products supporting IPv4 and IPv6 :

- CUCM/SME SIP Trunks
- Newer SCCP based Phones
- SIP based IOS gateways
- Cisco Unified Communications Manager Express and CUBE
- SCCP based VG224 Analog gateways and Router FXS ports,
- IOS MTPs
- SIP Trunks to Unity Connection
 Dual Stack (IPv4 and IPv6) configuration is recommended

 Dual Stack ANAT is recommended for SIP Trunks
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/ipv6/ipv6srnd.html
Centralized and Distributed PSTN
Centralized PSTN
PSTN
Voicemail CUBE CUBE

Conferencing
System System
H323
Unified CM MGCP
Session SIP
Management
Edition Cluster
 Centralized IP PSTN deployments are becoming increasingly popular as

savings can be made through bulk call minute deals with Service Providers
and by eliminating TDM circuits in branches
 Bear in mind that with no WAN connectivity no PSTN calls can be made or
received
 A number of analog PSTN ports are often provided in the branch for back up
and emergency outbound calls
 For emergency calls to the centralized PSTN, MLPP can be used to pre-empt
existing WAN calls if insufficient CAC bandwidth is available
Distributed PSTN
System System
H323
Unified CM MGCP
Session SIP
Management
Edition
Cluster
CUBE CUBE CUBE

PSTN PSTN PSTN PSTN PSTN PSTN
 Typically TDM based branch PSTN circuits

 Distributed IP PSTN deployments growing
 Enables Tail End Hop Off
 Can be combined with centralized outbound PSTN on SME or

centralized outbound PSTN on Leaf clusters which allows (for
example, in country) TEHO to be deployed
SME Cluster Sizing
SME Cluster Sizing – Summary Guidance
User Traffic information required to size your SME cluster
Average User Busy Hour Call Attempts (BHCA)
Average Call Holding Time (CHT)
% Off Net Traffic (Off Net to PSTN via SME)
% On Net Traffic
% Intra cluster Traffic
Work with your Cisco account team or Cisco partner to size your SME cluster using the
SME sizing tool
http://tools.cisco.com/cucst/faces/login.jsp
PSTN
Voicemail CUBE CUBE

Conferencing
System System
H323
Unified CM MGCP
Session SIP
Management
Edition Cluster
PBXs/ CUCM CUCM

CUCM Clusters with PBXs with PBXs with MGCP CUCM/PBXs with SIP
Clusters with Clusters with MGCPTrunks to SME Q931 Gateway QSIG Trunks to SME
SIP Trunks to SME H323 Annex M1
H323 QSIG Trunks Trunks to SME Trunks
to SME
Summary
Summary
 SME deployments allow you to simplify the UC network
edge by aggregating dial plan, services, applications
and PSTN access on an SME cluster
 SME based UC aggregation simplifies the management
of your UC network as it grows and changes
 For design guidance see the SME Deployment Guide :
 http://www.cisco.com/en/US/products/ps10661/product
s_implementation_design_guides_list.html
Additional Slides
SIP Trunk Signaling and Basic Operation
SIP Messaging - Delayed and Early Offer
SIP Early Offer
Information about the calling device’s media characteristics are sent with its initial SIP INVITE
message – The media characteristics are contained in the Session Description Protocol (SDP)
body sent with the SIP INVITE – The “Offer” in the SDP body will contain the IP Address, UDP
Port number, list of codecs etc. supported by the calling device
The called device selects which of the offered codecs it wishes to use for the call and returns its
“Answer” in the SDP body of a SIP response – The Answer also contains the IP address and
UDP port number etc of the called device
Once the Answer has been received and acknowledged two way media can be established
Early Offer is widely used …….

INVITE w/ SDP (Offer) INVITE w/ SDP (Offer)
200 OK w/ SDP (Answer) 200 OK w/ SDP (Answer)

Two Way Media
ACK ACK
SIP Delayed Offer
No information about the calling device’s media characteristics are sent with its initial SIP INVITE
message. Instead the media characteristics are sent by the called device in the Session
Description Protocol (SDP) body of the next reliable message (200 OK) – The called device’s
“Offer” will contain its IP Address, UDP Port number, list of codecs etc.
The calling device selects which of the offered codecs it wishes to use for the call and returns its
“Answer” in the SDP body of a reliable SIP response (ACK) – The Answer also contains the IP
address and UDP port number etc of the calling device
Delayed Offer is a mandatory part of the SIP standard – Many SPs prefer Early Offer
Ordinarily, the Offer or Answer cannot be sent with 100 Trying or 180 Ringing as 1XX messages
are unreliable (unacknowledged)… This can be resolved using PRACK ….. discussed later…..
INVITE INVITE
Two Way Media
Inbound SIP Delayed Offer to Outbound SIP Early offer
So what happens when Unified CM receives an inbound call on a Delayed Offer Trunk and
needs to onward route the call over a Early Offer Trunk ?
It does not have the calling device’s media characteristics and it needs to send an Offer in
SDP with the outbound INVITE…
Solution – Insert a Media Termination Point (MTP) and use its media characteristics to create
the Offer in SDP with the outbound INVITE
SIP Delayed Offer SIP Early Offer
INVITE INVITE w/ SDP (MTP)
200 OK w/ SDP (Offer) 200 OK w/ SDP (Answer)
ACK w/ SDP (Answer) ACK
Two Way Media

MTP
SIP Messaging – Enabling SIP Early Offer – Method 1
SIP Trunk “MTP Required” Checkbox
SIP Line SIP Trunk with Early Offer

MTP
SCCP Line SIP Trunk with Early Offer

MTP
SIP Trunk SIP Trunk with Early Offer

MTP
H323 Trunk SIP Trunk with Early Offer

MTP
MGCP Trunk SIP Trunk with Early Offer

MTP
Using the “MTP Required” option :
SIP Early Offer Trunks use the Trunk’s Media Termination Point (MTP) resources, inserting an
MTP into the media path for every outbound call – sending the MTP’s IP Address, UDP port
number and codec in the SDP body of the initial SIP INVITE instead of those of the endpoint.
This has a number of disadvantages : MTPs support a single Audio codec only e.g. G711 or
G729. The passthru codec is not supported excluding the use of SRTP and video calls. Since
the Trunk’s MTPs are used rather than the calling device’s MTPs - The media path is forced to
follow the signaling path.
New
SIP Trunk Signaling and Basic Operation Features
SIP Line SIP Trunk with Early Offer New SIP Profile checkbox “Early Offer
Cisco SIP Phones support for voice and video calls (insert
SCCP Line SIP Trunk with Early Offer MTP if needed)”
Newer SCCP Phones
SCCP Line SIP Trunk with Early Offer
For Calls from trunks and devices that can
MTP provide their IP Address, UDP port number
Older SCCP Phones
and supported codecs - This information is
SIP Trunk SIP Trunk with Early Offer
sent in the SDP body of the initial SIP Invite on
SIP Early Offer the outbound Early Offer Trunk. No MTP is
SIP Trunk SIP Trunk with Early Offer used for the Early Offer
MTP
SIP Delayed Offer

For Calls from trunks and devices that cannot
MTP provide Early Offer information – use the
H323 Slow Start
calling device’s MTP resources (first) or the
outbound trunk’s MTPs (second) to create a
H323 Fast Start SIP Offer for an unencrypted voice call. (SRTP
MGCP Trunk SIP Trunk with Early Offer and video can subsequently be initiated by the
MGCP Gateway called device)
New
SIP Trunk Signaling and Basic Operation Features

SIP Line SIP Trunk with Early Offer SIP Trunk SIP Trunk with Early Offer
MTP
Cisco SIP Phones SIP Delayed Offer
SCCP Line SIP Trunk with Early Offer H323 Trunk SIP Trunk with Early Offer
MTP
Newer SCCP Phones H323 Slow Start
SCCP Line SIP Trunk with Early Offer H323 Trunk SIP Trunk with Early Offer
MTP
Older SCCP Phones H323 Fast Start
SIP Trunk SIP Trunk with Early Offer MGCP Trunk SIP Trunk with Early Offer
SIP Early Offer MGCP Gateway
Benefits of “Early Offer support for voice and video calls (insert MTP if needed)”
• Reduced MTP usage

• Single voice codec MTP limitation removed (by using the pass through codec)
• Voice codecs sent in SIP Offer based on calling device capabilities & region settings
• Video Calls supported
• Encryption supported
• Use of the Calling device’s MTP rather than Trunk’s MTP
- media does not have to follow the signaling path
SIP Trunk Signaling and Operation – PRACK (1)
SIP Early Media – Using Provisional Acknowledgement (PRACK)
SIP defines two types of responses: Final and Provisional.
Final responses convey the result of the processed request, and are sent reliably (i.e. they are
acknowledged).
Provisional responses provide information on the progress of the request, but are not sent
reliably – so the sender of a provisional response does know that it has been received.
To send an Offer or Answer with a provisional 1XX response – these responses must be sent
reliably…..
PRACK – Provisional Acknowledgement is used to provide 1XX responses with reliability.
Diagram : Early Offer with Early Media
INVITE w/ SDP Supported:100rel INVITE w/ SDP Supported:100rel
183 Progress w/ SDP Require:100rel 183 Progress w/ SDP Require:100rel
PRACK Two Way Media PRACK
200 OK (PRACK) 200 OK (PRACK)
SIP Trunk Signaling and Operation – PRACK (2)
SIP Early Media – Using Provisional Acknowledgement (PRACK)
Like final responses, by using PRACK - 1XX messages will be periodically re-sent until their
receipt is acknowledged by the receiver by sending a PRACK, which is also acknowledged by
the 1XX sender.
Using PRACK can reduce the number of SIP messages that need to be sent before two way
media can be established
PRACK is useful in situations where long Round Trip Times between SIP devices can cause a
delay to media cut through or media clipping
PRACK can be enabled on the SIP Trunk Profile by setting “SIPRel1XX Options” to enabled
Diagram : Delayed Offer with Early Media
INVITE Supported:100rel INVITE Supported:100rel
183 Progress w/ SDP Require:100rel 183 Progress w/ SDP Require:100rel
PRACK w/ SDP PRACK w/ SDP
Two Way Media

200 OK (PRACK) 200 OK (PRACK)
Using standard Call Manager Groups and
multiple destinations IP addresses
A SIP Trunk D A SIP Trunk D
B E B E
C F C F
Cluster 1 Cluster 2 Cluster 1 Cluster 2
Unified CM SIP Trunks will only accept inbound calls from a device with an
IP address that has been defined as a destination IP address on the Trunk
Cluster 1 – SIP Trunk configuration Cluster 2 – SIP Trunk configuration
Servers A, B and C in SIP Trunk’s Servers D, E and F in SIP Trunk’s

Call Manager Group Call Manager Group
Servers D, E and F are configured Servers A, B and C are configured

as Trunk destinations as Trunk destinations
Using Run on all Active Unified CM Nodes and multiple
destination IP addresses
E E
A SIP Trunk A SIP Trunk
F F
B B
G G
C C
H H
D D
I I
Unified CM SIP Trunks will only accept inbound calls from a device with an IP
address that has been defined as a destination IP address on the Trunk
The SIP Trunk has an active SIP The SIP Trunk has an active SIP
daemon on Servers A, B, C and D daemon on Servers E, F, G, H and I
Servers E, F, G, H and I are Servers A, B, C and D are

configured as Trunk destinations configured as Trunk destinations
Using standard Call Manager Groups, Run on all Active
Unified CM Nodes and multiple destination IP addresses
A A
SIP Trunk SIP Trunk
B F B F
C G C G
D H D H
E E
Unified CM SIP Trunks will only accept inbound calls from a device with an IP
address that has been defined as a destination IP address on the Trunk
The SIP Trunk has an active SIP Servers F, G and H in SIP Trunk’s
daemon on Servers A, B, C, D and E Call Manager Group
Servers F, G and H are defined as Servers A, B, C, D and E are defined

Trunk destinations as Trunk destinations
Signaling Delay – SIP Delayed Offer Trunks

caller hears ringback tone

INVITE INVITE
SME
Two Way Media
Delay before the caller hears the Messages exchanged before

called user after ringback stops called user hears the caller after
picking up their handset
Signaling Delay – SIP Early Offer Trunks

caller hears ringback tone
INVITE w/ SDP (Offer) INVITE w/ SDP (Offer)

200 OK w/ SDP (Answer) 200 OK w/ SDP (Answer)
ACK ACK
Two Way Media
Delay before the caller hears the Messages exchanged before

called user after ringback stops called user hears the caller after
picking up their handset
Recommended Reading
BRKUCC- 2931
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Designing &amp; Deploying UC Networks

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Designing & Deploying UC networks with Cisco

Session Management Edition

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 SME DEPLOYMENT GUIDE

A CUCM cluster is typically used to register 10,000s of Phones

Support for SME deployments was introduced with UC version 7.1(2)

UC version 8.5 introduces a number of features that enhance SME functionality :

Improved SIP Trunk and H323 Inter Cluster Trunk functionality

Improved through-cluster routing

SIP Normalization and Transparency scripts

CUCM PBXs/ CUCM CUCM PBXs with PBXs with CUCM/PBX

Leaf Unified CM Clusters/ Leaf UC Systems

An SME cluster can interconnect 1000’s of UC systems using SIP, H323, or

Voicemail CUBE CUBE Conferencing

PBXs/ CUCM CUCM

A centralized Dial Plan in SME allows you to :

PSTN PSTN PSTN PSTN PSTN PSTN

Phase 1 – PBX Phase 2 – Centralized IP

Phase 3 – PBX Phase 4 – CUCM cluster

 Summary deployment recommendations for SME Trunks

 UC version 8.5+ is recommended for SME clusters as it introduces a range

 Although MGCP Trunks to gateways offer benefits in terms of ease of dial

BRKUCC-2931 Legend: Yes Limited support

 H323 Inter Cluster Trunks

 SIP Trunks and SIP Inter Cluster Trunks

In conjunction with the “Route Local” feature – these new features

 Run On All Unified CM Nodes

CUCM SME CUCM SME

OPTIONS Ping is activated on a per SIP Trunk basis

- CUCM will not attempt to establish a call to an unavailable remote peer

- CUCM 8.5 SIP Trunks – Dynamic reachability detection

HA Features affecting Calls originating from HA Features for Trunk

Calls over single Trunks Calls over multiple Trunks

Route Lists and Route Up to 16 Destination IP

Call Manager Groups Call Manager Groups H323 Gatekeeper

Run on All Nodes Run on All Nodes SIP OPTIONS Ping

Route Local Route Local

SIP Trunk SIP Trunk IP PSTN The Normalization feature is designed to

Script Normalization is independent of what the

Normalization Script • SIP SIP Trunk calls

SIP Trunk SIP Trunk The Transparency feature is designed

H323 Trunk B MGCP Trunk B

Route List Route List

H.323 SIP MGCP

Hook-flash Transfer with Unified CM

TDM Variations: A-DID, E&M, PRI NFAS, CAMA, T1 FGD

SIP Early Offer CUBE SIP DO to EO

SIP ICT Trunks - Voice, Video and Encryption supported

East Coast DC SIP

In each Leaf Cluster :

East Coast DC SIP

In the SME Cluster :

Call Progress failure messages (e.g. No circuit, Number Unavailable,

 Secure Signaling and encrypted Media can be configured

SIP Inter Cluster Trunk using

In the SME cluster a unique globalized dial plan should be

In Leaf clusters/ Leaf UC systems a localized dial plan

With Session Management Edition and CUCM, normalisation of calling and

• Transformation calling search spaces on trunk or device pool level

This presentation focuses on number transformations at the Trunk level......

Cluster 1 DNs Cluster 2 DNs

Calling Number +44-404-444-4000 Calling Number +44-404-444-4000

Caller (444-4000) dials 8-555-5000

Designing & Deploying UC Networks

Designing & Deploying UC Networks