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Americas Headquarters

Cisco Systems, Inc.170 West Tasman DriveSan Jose, CA 95134-1706USAhttp://www.cisco.comTel: 408 526-4000

800 553-NETS (6387)Fax: 408 527-0883

Cisco Unified Communications ManagerSession Management Edition, Release

8.6(1)Version 8.6(1)

Text Part Number: OL-25265-01
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Any Internet Protocol (IP) addresses and phone numbers used in this document are not i ntended to be actual addresses and phone numbers. Any examples, command display

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Cisco Unified Communications Manager Session Management Edition, Release 8.6(1) 2011 Cisco Systems, Inc. All rights reserved.
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C O N T E N T S

Preface v

Overview 1-1

Unified CM Session Management Edition Basics 1-1

When to Deploy Unified CM Session Management Edition 1-2

Differences Between Unified CM Session Management Edition and Standard Cisco Unified CMClusters 1-3

Architecture Overview 1-4

SIP Trunks 1-4

SIP Trunks for IP PSTN Connectivity 1-5H.323 Intercluster Trunks 1-5

H.323 Gateways 1-6

MGCP Gateways 1-6

Features New to Cisco Unified CM Release 8.5(1) 1-6

Trunk Type and Feature Recommendations for Multi-Cluster Deployments 1-7

Multiple Clusters That Run Cisco Unified CM 8.5 1-7

Multiple Clusters That Run Cisco Unified CM 8.5 and Pre-8.5 Versions 1-8

Unified CM Session Management Edition Components 1-10

Unified CM Session Management Edition Clusters 1-10

Unified CM Leaf Clusters 1-11

Third Party Leaf Cisco Unified CM Systems 1-11

IP PSTN Connectivity 1-11

Centralized Applications 1-11

Before you Begin 1-12

Licensing 1-12

Interoperability Testing 1-12

Installation Considerations 1-13

Unified CM Session Management Edition Design Considerations 1-13

Cisco Unified Communications Manager Trunks 2-1

A Comparison of SIP and H.323 Trunks 2-1

SIP Trunks Overview 2-4

General Deployment Considerations 2-5

SIP Trunk Features and Operation 2-5

SIP Trunks Can Run on All Active Unified CM Nodes 2-5

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Up to 16 SIP Trunk Destination IP Addresses 2-5

SIP OPTIONS Ping 2-6

SIP Early Offer Support over Cisco Unified CM SIP Trunks 2-6

QSIG over SIP Trunks 2-9

SIP Trunk Message Normalization and Transparency 2-9

Route Lists Run on All Active Unified CM Nodes 2-13

SIP Trunks Using DNS 2-14

High Availability for SIP Trunks 2-16

Multiple Source Cisco Unified CM Servers for Originating SIP Trunk Calls 2-16

Multiple Destination IP Addresses per SIP Trunk 2-17

Design Considerations When Using Run on All Active Unified CM Nodes 2-17

Multiple SIP Trunks Using Route Lists and Route Groups 2-17


Load Balancing for SIP Trunks 2-17Outbound Calls over a Single SIP Trunk 2-18

Outbound Calls over Multiple SIP Trunks 2-18


SIP Delayed Offer and Early Offer 2-18

Media Termination Points 2-19

DTMF Transport 2-21

SIP Trunk Transport Protocols 2-22

Secure SIP Trunks 2-22

Media Encryption 2-22

Signaling Encryption 2-22

Calling Party Number Transformation and SIP Trunks 2-23

SIP Trunk Service Types 2-24

Design Considerations for SIP Trunks 2-25

Considerations for SIP Intercluster Trunks 2-25

Using Standard Cisco Unified CM Groups with SIP Intercluster Trunks 2-25

Using Run on All Active Unified CM Nodes with SIP Intercluster Trunks 2-26

Using Standard Cisco Unified CM Groups and Run on All Active Unified CM Nodes with SIPIntercluster Trunks 2-27

Trunk Type and Feature Recommendations for Multi-Cluster Deployments 2-28

Multiple Clusters All Running Cisco Unified CM 8.5 or Later Releases 2-29

Multiple Clusters Running Cisco Unified CM 8.5 and Prior Releases 2-30

Trunk Design Considerations for Clustering over the WAN 2-31

Design Guidance for Clustering over the WAN with Leaf Cluster Trunks 2-32

Design Guidance for Clustering over the WAN with Unified CM Session Management EditionCluster Trunks 2-33

Other SIP Trunk Deployment Considerations 2-33

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H.323 Trunks Overview 2-34

General H.323 Intercluster Trunk Deployment Considerations 2-34

Basic Operation of H.323 Trunks 2-35

H.323 Trunk Types 2-35

Intercluster Trunk (Non-Gatekeeper Controlled) 2-35

Intercluster Trunk (Gatekeeper Controlled) 2-42

H.225 Trunk (Gatekeeper Controlled) 2-43

High Availability for Gatekeeper Controlled Trunks 2-43

Load Balancing Outbound Calls over H.323 Gatekeeper Controlled Trunks 2-46

H.323 Outbound FastStart Call Connections 2-47

H.323 Trunks with Media Termination Points 2-48

DTMF Transport 2-48

H.323 Trunk Transport Protocols 2-48

Secure H.323 Trunks 2-48H.323 Operation in Cisco Unified CM 2-49

Other Design Considerations for H.323 Trunks 2-53

General SIP and H.323 Trunk Design Considerations 2-53

Deterministic Outbound Call Load Balancing over Cisco Unified CM Trunks 2-53

Codec Selection Over IP Trunks 2-54

Other MTP Uses 2-55

Other Design Considerations for Unified CM Session Management Edition Deployments 2-55

Unified CM Session Management Edition and SAF CCD Deployments 2-56

Advertising SAF CCD Routes to Leaf Clusters from/through Unified CM Session Management

Edition 2-56

Advertising SAF CCD Routes from Leaf Clusters to Unified CM Session Management Edition 2-56

Operational Considerations for Unified CM Session Management Edition and SAF CCDDeployments 2-58

Cisco Unified CM Trunks and Emergency Services 2-59

Capacity Planning for Cisco Unified CM IP Trunks 2-59

IP PSTN and IP Trunks to Service Provider Networks 2-60

Cisco Unified Border Element 2-60

Protocol and Media Interworking 2-60

Service Provider SIP Trunk Best Practice Recommendations 2-61

Design Considerations for Scalability and Redundancy 2-61

Security Best Practices 2-62

Management and Monitoring of Service Provider SIP Trunks 2-63

Trunk IP-PSTN Connection Models 2-63

Deployment Examples A-1

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Preface

This preface describes the purpose, audience, organization, and conventions of this guide and provides

information on how to obtain related documentation.

Note This document may not represent the latest Cisco product information available. You can obtain the most

current documentation by accessing Cisco's product documentation page at this URL:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/tsd_products_support_series_home.html

The preface covers these topics:

Purpose, page v

Audience, page v

Organization, page vi

Related Documentation, page vi

Conventions, page vi

Obtaining Documentation and Submitting a Service Request, page vii

Cisco Product Security Overview, page viii

PurposeThis document provides conceptual information about Cisco Unified Communicat ions Manager Session

Management Edition and its components as well as design considerations and guidelines for deploying

Cisco Unified Communications Manager Session Management Edition.

AudienceThe Cisco Unified Communications Manager Session Management Edition Deployment Guide provides

information for network administrators and designers who are responsible for deploying the Cisco

Unified Communications Manager Session Management Edition system. This guide requires knowledge

of telephony and IP networking technology.
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OrganizationThe following table shows the organization of this guide:

Related DocumentationFor information about Cisco Unified Communications Manager documentation, refer to the

documentation guide for your release at the following URL:

http://cisco.com/en/US/products/sw/voicesw/ps556/products_documentation_roadmaps_list.html .

ConventionsThis document uses the following conventions:

Chapter DescriptionChapter 1 Overview

This chapter provides an overview of Cisco Unified Communications Manager

Session Management Edition and a description of the system components.

Chapter 2 Cisco Unified Communications Manager Trunks

This chapter provides information about the trunks used in a Cisco Unified

Communications Manager Session Management Edition deployment as well as

design considerations for the system.

Appendix A Deployment Examples

This appendix provides a link to Cisco Unified Communications Manager Session

Management Edition deployment models.

Convention Description

boldface font Commands and keywords are in boldface.

italic font Arguments for which you supply values are in italics.

[ ] Elements in square brackets are optional.

{ x | y | z } Alternative keywords are grouped in braces and separated by vertical bars.

[ x | y | z ] Optional alternative keywords are grouped in brackets and separated by

vertical bars.

string A nonquoted set of characters. Do not use quotation marks around the stringor the string will include the quotation marks.

screen font Terminal sessions and information the system displays are in screen font.

boldface screen font Information you must enter is inboldface screen font.

italic screen font Arguments for which you supply values are in italic screen font.

This pointer highlights an important line of text in an example.
http://cisco.com/en/US/products/sw/voicesw/ps556/products_documentation_roadmaps_list.htmlhttp://cisco.com/en/US/products/sw/voicesw/ps556/products_documentation_roadmaps_list.html

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Notes use the following conventions:

Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the

publication.

Timesavers use the following conventions:

Timesaver Means the described action saves time. You can save time by performing the action described in the

paragraph.

Tips use the following conventions:

Tip Meansthe information contains useful tips.

Cautions use the following conventions:

Caution Means reader be careful. In this situation, you might do something that could result in equipment

damage or loss of data.

Warnings use the following conventions:

Warning This warning symbol means danger. You are in a situation that could cause bodily injury. Before you

work on any equipment, you must be aware of the hazards involved with electrical circuitry and

familiar with standard practices for preventing accidents.

Obtaining Documentation and Submitting a Service RequestFor information on obtaining documentation, submitting a service request, and gathering additional

information, see the monthly Whats New in Cisco Product Documentation, which also lists all new and

revised Cisco technical documentation, at:

http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html

Subscribe to the Whats New in Cisco Product Documentation as a Really Simple Syndication (RSS) feed

and set content to be delivered directly to your desktop using a reader application. The RSS feeds are a free

service and Cisco currently supports RSS Version 2.0.

^ The symbol represents the key labeled Controlfor example, the key

combination ^D in a screen display means hold down the Control key while

you press the D key.

< > Nonprinting characters, such as passwords, are in angle brackets.

Convention Description
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Cisco Product Security OverviewThis product contains cryptographic features and is subject to United States and local country laws

governing import, export, transfer and use. Delivery of Cisco cryptographic products does not imply

third-party authority to import, export, distribute or use encryption. Importers, exporters, distributors

and users are responsible for compliance with U.S. and local country laws. By using this product youagree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local

laws, return this product immediately.

Further information regarding U.S. export regulations may be found at

http://www.access.gpo.gov/bis/ear/ear_data.html .
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C H A P T E R

1-1


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1Overview

Unified Communications deployments using Cisco Unified Communications Manager Session

Management Edition (Unified CM Session Management Edition) are a variation of the multisite

distributed call processing deployment model and are typically employed to interconnect large numbers

of unified communications systems through a single front-end system, in this case the Unified CM

Session Management Edition. This document discusses the relevant design considerations for deploying

Unified CM Session Management Edition.

Unified CM Session Management Edition is essentially a Cisco Unified Communications Manager

(Cisco Unified CM) cluster with trunk interfaces only and no IP endpoints. It enables aggregation of

multiple unified communications systems, referred to as leaf systems.

Unified CM Session Management Edition deployments can be used to migrate a deployment of multiple

PBXs and associated phones to a Cisco Unified CM cluster with IP phones and relatively few trunks.

The Unified CM Session Management Edition cluster may start with a large number of trunks

interconnecting third-party PBXs; and migrate over time to a Cisco Unified CM cluster deployment with

thousands of IP phones.

This section contains the following information:

Unified CM Session Management Edition Basics, page 1-1

Architecture Overview, page 1-4

Trunk Type and Feature Recommendations for Multi-Cluster Deployments, page 1-7

Unified CM Session Management Edition Components, page 1-10

Centralized Applications, page 1-11

Before you Begin, page 1-12

Unified CM Session Management Edition BasicsWith Cisco Unified CM 8.0 and later releases, Unified CM Session Management Edition supports the

following features:

H.323 Annex M1 intercluster trunks

SIP intercluster trunks

SIP trunks

H.323 trunks

MGCP trunks

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Unified CM Session Management Edition Basics

systems with a simplified dial plan and trunk(s) that point to the session management cluster.

Unified CM Session Management Edition maintains the centralized dial plan and corresponding

reachability information about all of the other unified communications systems.

Provide centralized PSTN accessUnified CM Session Management Edition can be used to

aggregate PSTN access to one (or more) centralized IP PSTN trunks. Centralized PSTN access is

commonly combined with the reduction, or elimination, of branch-based PSTN circuits. Centralize applicationsThe deployment of a Unified CM Session Management Edition enables

commonly used applications, such as conferencing or videoconferencing to connect directly to the

session management cluster, which reduces the overhead of managing multiple trunks to leaf

systems.

Aggregate PBXs for migration to a unified communications systemUnified CM Session

Management Edition provides an aggregation point for multiple PBXs as part of the migration from

legacy PBXs to a Cisco Unified CM system.

Differences Between Unified CM Session Management Edition and Standard

Cisco Unified CM ClustersThe Unified CM Session Management Edition software is exactly the same as Cisco Unified CM;

however, the software has been enhanced significantly to satisfy the requirements and the constraints of

this new deployment model. Unified CM Session Management Edition is designed to support a large

number of trunk-to-trunk connections, and as such, it is subject to the following design considerations:

CapacityIt is important to correctly size the Unified CM Session Management Edition cluster

based on the expected BHCA traffic load between leaf unified communications systems (for

example, Cisco Unified CM clusters and PBXs), to and from any centralized IP PSTN connections,

and to any centralized applications. Work with your Cisco account Systems Engineer (SE) or Cisco

partner to size your Unified CM Session Management Edition cluster correctly.

TrunksWhere possible, avoid the use of static MTPs on Cisco Unified CM trunks (that is, "MTP

required" must not be checked on the leaf or session management Cisco Unified CM SIP or H.323trunks). MTP-less trunks offer more codec choices, support voice, video, and encryption, and do not

anchor trunk calls to MTP resources. If SIP Early Offer is required by a third-party unified

communications system, use the Delayed Offer to Early Offer feature with Cisco Unified Border

Element (CUBE). Dynamically inserted MTPs can be used on trunks (for example, for DTMF

translation from in-band to out-of-band).

Unified CM versions Both the Unified CM Session Management Edition and Cisco Unified CM

leaf clusters should be deployed with Cisco Unified CM 7.1(2) or later release. Earlier versions of

Cisco Unified CM can be deployed but might experience problems that can be resolved only by

upgrading your cluster to a Cisco Unified CM 7.1(2) or later release.

InteroperabilityThough most vendors do conform to standards, differences can and do exist

between protocol implementations from various vendors. As with any standard Cisco Unified CM

cluster, Cisco strongly recommends that you conduct end-to-end system interoperability testingwith any unverified third-party unified communications system before deploying the system in a

production environment. The interoperability testing should verify call flows and features from

Cisco and third-party leaf systems through the Unified CM Session Management Edition cluster. To

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Architecture Overview

learn which third-party unified communications systems have been tested by the Cisco

Interoperability team, go to www.cisco.com/go/interoperability and select the link for Cisco Unified

Communications Manager - Session Management Edition.

Load balancing for inbound and outbound callsConfigure trunks on the Unified CM Session

Management Edition and leaf unified communications systems so that inbound and outbound calls

are evenly distributed across the Cisco Unified CM servers within the session management cluster.For more information on load balancing for trunk calls, refer to the chapter on Cisco Unified

Communications Manager Trunks section on page 2-1.

Design assistanceUnified CM Session Management Edition designs should be reviewed by your

Cisco SE in conjunction with the Cisco Unified CM Session Management Team. For more

information of the Unified CM Session Management Edition design review process, Cisco partners

and employees can refer to the documentation at the following URL:

http://docwiki.cisco.com/wiki/Unified_Communications_Manager_-_Session_Manager_Edition

Architecture OverviewEvery Unified CM Session Management Edition design consists of a Unified CM Session Management

Edition cluster and a number of leaf Cisco Unified CM clusters or leaf third-party unified

communications systems. How these leaf systems connect to the Unified CM Session Management

Edition cluster largely depends on the protocols and features that are required to be supported by each

system. A Unified CM Session Management Edition cluster can support SIP, H.323 or MGCP Trunks.

Each of these trunk protocols have their own merits, but as a general rule, with a Unified CM Session

Management Edition cluster that runs Cisco Unified CM 8.5, Cisco recommends SIP trunks as they offer

a greater set of features when compared to H.323 and MGCP trunks. The choice of trunk used on Unified

CM Session Management Edition is often determined by the end system to which Unified CM Session

Management Edition is connecting.

The following topics give a brief overview of Unified CM Session Management Edition trunk

capabilities (the features and capabilities of Unified CM Session Management Edition trunk types are

discussed in detail later in this document) and decribes a number of Unified CM Session ManagementEdition trunking use cases:

SIP Trunks, page 1-4

H.323 Intercluster Trunks, page 1-5

H.323 Gateways, page 1-6

MGCP Gateways, page 1-6

SIP Trunks

SIP trunks can be used as intercluster trunks between Unified CM Session Management Edition and leaf

Cisco Unified CM clusters, as trunks to third party unified communications systems, and SIP-based IP

PSTNs. Today, SIP is arguably the most commonly chosen protocol when connecting to service

providers and unified communications applications. Cisco Unified CM 8.5 provides the following SIP

trunk and call routing related features and functionality, which provides highly available trunk

connections with dynamic reachability detection and good load balancing capabilties :

Up to 16 destination IP addresses per trunk, which can be run on all Cisco Unified CM nodes

SIP OPTIONS Ping keepalives
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SIP Early Offer support for voice and video calls (insert MTP if needed)

QSIG over SIP

SIP Trunk Normalization and Transparency

Route Lists that can be run on all Unified CM nodes

Because of the SIP trunk features that are available in this release, Cisco recommends SIP for new andexisting trunk connections.

SIP Trunks for IP PSTN Connectivity

Service providers are increasingly offering IP PSTN connections to enterprise customers. Cisco strongly

recommends that you connect to the IP PSTN via a CUBE as an enterprise edge session border controller.

(SBC). CUBE provides a controlled demarcation and security point for the following functions:

Address and port translations (privacy and Layer 7 topology hiding)

Media interworking (DTMF, fax, codec transcoding and transrating, volume/gain control,

RTP-SRTP conversion)

Call Admission Control (based on total calls, CPU, Memory, call arrival spike detection, ormaximum calls per destination)

Security (DoS attack protection, toll fraud protection, SIP malformed packet detection, and number

of signaling and media encryption options)

Complaince with the SP SIP trunk UNI specification

H.323 Intercluster Trunks

For intercluster trunk (ICT) connections between Unified CM Session Management Edition and Cisco

Unified CM leaf clusters, where the leaf cluster is using a version of Cisco Unified CM prior to release

8.5, Cisco generally recommends H.323 non-gatekeeper-controlled ICTs as they offer addditional

features in comparison with SIP and MGCP trunks, such as up to three configurable destination

addresses, support for H.323 Annex M1 (QSIG over H.323).

For ICT connections between Unified CM Session Management Edition and Cisco Unified CM leaf

clusters using Cisco Unified CM release 8.5, H.323 non-gatekeeper-controlled ICTs offer the following

additional functionality:

Run on all Cisco Unified CM nodes

Allow up to 16 destination IP addresses per trunk

Run route lists on all Cisco Unified CM nodes

Note H.323 gatekeeper-controlled ICTs do not support the capability to run on all Cisco Unified CM nodes

or allow up to 16 destination IP addresses to be defined per trunk; however, they can support more than16 destination IP addresses by using the gatekeeper to return destination addresses to the calling cluster

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H.323 Gateways

Use H.323 gateway trunks for non-gatekeeper-controlled connections to H.323-based third party Cisco

Unified CM systems. H.323 gateway trunks support standard Cisco Unified CM groups and a single

configurable destination IP address.

MGCP Gateways

MGCP gateways support standard Cisco Unified CM groups and a single configurable destination IP

address. Prior to Cisco Unified CM 8.5, MGCP gateways were the only option for connecting a

QSIG-based TDM PBX to Cisco Unified CM via an IOS gateway. Cisco Unified CM 8.5 now supports

QSIG over SIP connections to IOS gateways. These standard Cisco Unified CM SIP trunks support all

of the features described previously for SIP trunks. MCGP trunks are often preferred by customers

because minimal configuration is required in the IOS gateway. For example, no dial peers need to be

configured in the IOS gateway as the Q.931/QSIG signaling is backhauled to Cisco Unified CM.

Although MGCP trunks reduce configuration overhead, they do not support the dynamic reachability

detection and load balancing capabilties of SIP trunks.

Features New to Cisco Unified CM Release 8.5(1)

Cisco Unified CM 8.5 and later releases provide the following SIP trunk and call routing enhancements.

These enhancements can be utilized in a Unified CM Session Management Edition deployment when the

system components run this version of Cisco Unified CM.

SIP OPTIONSThe SIP OPTIONS method allows a SIP trunk to track the status of remote

destinations. By sending outgoing OPTIONS and checking the incoming response message, the SIP

trunk knows whether remote peers are ready to receive a new request. The SIP trunk does not attempt

to set up new calls to unreachable remote peers; this approach allows for quick failovers.

Up to 16 destination IP addresses per trunkFor SIP trunks, Cisco Unified CommunicationsManager supports up to 16 IP addresses for each DNS SRV and up to 10 IP addresses for each DNS

host name. The order of the IP addresses depends on the DNS response and may be identical in each

DNS query. The OPTIONS request may go to a different set of remote destinations each time if a

DNS SRV record (configured on the SIP trunk) resolves to more than 16 IP addresses, or if a host

name (configured on the SIP trunk) resolves to more than 10 IP addresses. Thus, the status of a SIP

trunk may change because of a change in the way a DNS query gets resolved, not because of any

change in the status of any of the remote destinations.

SIP Early Offer support for voice and video calls (insert MTP if needed)To enhance

interoperability with third party SIP devices, Cisco Unified CM now allows you to configure SIP

trunks to enable early offer for outgoing voice and video calls without requiring MTP, when media

capabilities and media port information of the calling endpoint is available. For outgoing call setup

for an early offer trunk, Cisco Unified CM includes an SDP with the calling device media port,

codecs, and IP address of the calling device (when available); inserts an MTP for early offer only

when the media information for the caller is unavailable; and advertises multiple codecs when an

MTP that supports multiple codecs gets inserted. The early offer feature enhancement in this release

ensures that a higher percentage of outbound early offer SIP trunk calls get made without requiring

an MTP, thus reducing the number of MTP resources needed and improving interoperability with

third party PBXs.

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Trunk Type and Feature Recommendations for Multi-Cluster Deployments

QSIG over SIPCisco Unified CM supports QSIG tunneling over a SIP trunk. QSIG tunneling

supports the following features: Call Back, Call Completion, Call Diversion, Call Transfer,

Identification Services, Message Waiting Indication, and Path Replacement.

SIP trunk normalization and transparencySIP transparency and normalization allow Cisco

Unified CM to seamlessly interoperate with a variety of PBXs and service providers. Normalization

allows you to modify incoming and outgoing SIP messages at a protocol level on their way throughCisco Unified CM. Transparency allows Cisco Unified CM to pass headers, parameters, and content

bodies from one call leg to the other.

Run route lists on all Cisco Unified CM nodes The SIP trunk features available in the 8.5 release

make SIP the preferred choice for new and existing trunk connections. The QSIG over SIP feature

provides parity with H.323 intercluster trunks and can also be used to provide QSIG over SIP trunk

connections to Cisco IOS gateways (and on to QSIG-based TDM PBXs). The ability to run on all

Cisco Unified CM nodes and to handle up to 16 destination IP addresses improves outbound call

distribution from Cisco Unified CM clusters and reduces the number of SIP trunks required between

clusters and devices. SIP OPTIONS ping provides dynamic reachability detection for SIP trunk

destinations, rather than per-call reachability determination. SIP Early Offer support for voice and

video calls (insert MTP if needed) can reduce or eliminate the need to use MTPs and allows voice,

video, and encrypted calls to be made over SIP Early Offer trunks.

For the complete list of new enhancements for SIP trunks for this release, refer to theNew and Changed

for Cisco Unified Communications Manager 8.5(1) document available at

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html

Trunk Type and Feature Recommendations for Multi-ClusterDeployments

This section describes the following multi-cluster deployments:

Multiple Clusters That Run Cisco Unified CM 8.5, page 1-7

Multiple Clusters That Run Cisco Unified CM 8.5 and Pre-8.5 Versions, page 1-8

Multiple Clusters That Run Cisco Unified CM 8.5

When all Cisco Unified CM clusters run version 8.5, the following SIP trunk features and functionality

should be used where applicable:

SIP OPTIONS Ping

Early offer support for voice and video (insert MTP if needed)

Capability to run on all nodes

Multiple destination IP addresses

QSIG over SIP

Deploying these features reduces MTP usage, provides high availability, even call distribution, and

dynamic SIP trunk failure detection. For inbound SIP trunk calls to Cisco Unified CM, Cisco

recommends SIP Early Offer.

SIP ICTs support voice, video, and encrypted media between Cisco Unified CM clusters, as well as all

of the features previously listed. If multiple trunks are used with route lists, enable the "Run on All

Nodes" feature on the route list.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.htmlhttp://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html

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For SIP trunks to an IP PSTN, SIP Early Offer is typically required by the service provider and most

providers support voice calls only; however, video calls and encrypted media are also supported. For

inbound SIP trunk calls to Cisco Unified CM, Cisco recommends SIP Early Offer.

SIP trunks to third party Cisco Unified CM systems may support voice, video, and encrypted media.

Check the capabilities of the end system to determine the SIP trunk features and media capabilities that

it supports. For inbound SIP trunk calls to Cisco Unified CM, Cisco recommends SIP Early Offer.

Note Be aware that SIP trunks on IOS gateways always send SIP Early Offer.

For SIP trunk connections to the IP PSTN and third party Cisco Unified CM systems, you can use

Normalization and Transparency scripts to address SIP interoperability issues. Cisco also recommends

the deployment of the CUBE on any IP PSTN SIP trunk connection from Cisco Unified CM to a voice

service provider. Figure 1-2 provides an example of multi-cluster deployments with Cisco Unified

CM 8.5.

Figure 1-2 Multi-Cluster Deployments With Cisco Unified CM 8.5

Multiple Clusters That Run Cisco Unified CM 8.5 and Pre-8.5 Versions

This section describes the trunk types and features that you should use when Cisco Unified CM leaf

clusters run release 8.5 and pre-8.5 versions of Cisco Unified CM.

When the leaf cluster runs an earlier version (pre-8.5) of Cisco Unified CM and voice, video and

encryption are required, use H.323 Slow Start ICTs and Annex M1 (QSIG), if desired. Deploy one or

more H.323 Slow Start ICTs that use standard Cisco Unified CM groups and up to three destination IP

addresses. If multiple trunks are used with route lists, and to avoid the route local rule described in the

Route Lists Run on All Active Unified CM Nodes section on page 2-13, ensure that the primary server

in the route list's Cisco Unified CM group is not co-resident on the same node as an associated outbound

H.323 trunk.

254879

SIP Early Offer

SIP Early Offer (insert MTP if needed)

Unified CM 8.5

IP

IP

IP PSTN

IP

IP

CUBE CUBESIP

Unified CM 8.5

Unified CMSession Management

Edition 8.5

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For leaf clusters that run Cisco Unified CM 8.5, use a SIP Delayed Offer ICT, enable the Run on All

Nodes feature, and use multiple destination IP addresses and SIP OPTIONS Ping for high availability

and even call distribution. If multiple trunks are used with route lists, enable the "Run on All Nodes"

feature on the route list.

Using SIP Delayed Offer ICTs on Cisco Unified CM 8.5 leaf clusters and H.323 Slow Start ICTs on leaf

clusters using earlier versions of Cisco Unified CM, allows voice, video and encrypted calls to be madebetween clusters and reduces the number of MTPs required (MTPs are only inserted when required for

DTMF translation, transcoding and so on).

For Unified CM Session Management Edition SIP trunks to IP PSTN deployments, SIP Early Offer is

typically required by the service provider and most providers support voice calls only. Use Early Offer

support for voice and video (insert MTP if needed), or Delayed Offer on this SIP trunk and, if supported

by the end system, SIP OPTIONS Ping, Run on All Nodes, and multiple destination IP addresses. For

inbound SIP trunk calls to Cisco Unified CM, Cisco recommends SIP Early Offer.

Unified CM Session Management Edition SIP trunks to third party Cisco Unified CM systems may

support voice, video and encrypted media. Use Early Offer support for voice and video (insert MTP if

needed), or Delayed Offer on this SIP trunk. Check the capabil ities of the end system to determine which

other Cisco Unified CM SIP trunk features are available. For inbound SIP trunk calls to Cisco Unified

CM, Cisco recommends SIP Early Offer.

Note Be aware that SIP trunks on IOS gateways always send SIP Early Offer.

For Unified CM Session Management Edition SIP trunk connections to the IP PSTN and third party

Cisco Unified CM systems, you can use Normalization and Transparency scripts to address SIP

interoperability issues. Cisco also recommends the deployment of the CUBE on any IP PSTN SIP trunk

connection from Cisco Unified CM to a voice service provider.

Figure 1-3 provides an example of multi-cluster deployments with Unified CM Session Management

Edition 8.5 and Cisco Unified CM leaf clusters that run release 8.5 and pre-8.5 versions of Cisco Unified

CM.

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Unified CM Session Management Edition Components

Figure 1-3 Multi-Cluster Deployments With Unified CM Session Management Edition 8.5, Cisco Unified CM 8.5 and

Cisco Unified CM Pre-8.5

Unified CM Session Management Edition ComponentsThis section describes the components of the Unified CM Session Management Editionand contains the

following topics:

Unified CM Session Management Edition Clusters, page 1-10 Unified CM Leaf Clusters, page 1-11

Third Party Leaf Cisco Unified CM Systems, page 1-11

IP PSTN Connectivity, page 1-11

Unified CM Session Management Edition Clusters

Unified CM Session Management Edition cluster servers can be deployed on the following platforms:

UCS B200 M1/M2 blade server

UCS C210 M1/M2 2U rack server

MCS 7845-I3 and HP DL380G6

As a general recommendation, Cisco Unified CM 8.5 or later should be deployed on the Unified CM

Session Management Edition cluster, as these versions benefit from a number of trunk-specific feature

enhancements. If you want to use an earlier version of Cisco Unified CM, Cisco recommends Cisco

Unified CM 7.1(2) or later. Although you can deploy versions of Cisco Unified CM that are earlier than

release 7.1(2), be aware that you may experience problems that can be resolved only by upgrading your

cluster to a Cisco Unified CM 7.1(2) or later release.

254880

SIP Early Offer

SIP Early Offer (insert MTP if needed)

Unified CM 8.0

IP

IP

IP PSTN

IP

IP

CUBE CUBESIP

Unified CM 8.5

Unified CMSession Management

Edition 8.5

H.323 Slow Start

H.323 Slow Start

SIP Delayed Offer

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Chapter 1 Overview

Centralized Applications

Unified CM Leaf Clusters

Cisco recommends that you deploy Cisco Unified CM leaf clusters with a Cisco Unified CM 7.1(2) or

later release. Earlier versions of Cisco Unified CM can be deployed but might experience problems that

can be resolved only by upgrading your cluster to Cisco Unified CM 7.1(2) or later release.

Third Party Leaf Cisco Unified CM Systems

In most cases, the third party Cisco Unified CM system's capabilities (for example, support for SIP

and/or H.323) determines the trunking protocol that is used by Unified CM Session Management

Edition. When an application supports both SIP and H.323 protocols, use SIP trunks with Unified CM

Session Management Edition 8.5 clusters. For Unified CM Session Management Edition clusters that

run versions prior to version 8.5, use either SIP trunks or H.323-based gateways/trunks.

Where possible, if SIP is used, the third party Cisco Unified CM system should send SIP Early Offer

over the SIP trunk to Unified CM Session Management Edition. If H.323 is used, the third party Cisco

Unified CM system should send H.323 Fast Start over the H.323 trunk to Unified CM Session

Management Edition.

IP PSTN Connectivity

CUBE is a key component in the enterprise network and is deployed at the edge of the enterprise network

for IP PSTN connectivity to service providers. It provides a controlled demarcation and security

network-to-network interface point. CUBE is a licensed Cisco IOS application that is available on the

following platforms:

Cisco Integrated Service Routers Generation 2 (ISR-G2s such as 29xx, 39xx)

Cisco 1000 Series Aggregation Services Routers (ASRs)

Cisco Integrated Service Routers (ISRs such as 28xx, 38xx)

Cisco AS5000XM Media Gateways

Depending on the hardware platform, CUBE can provide session scalability from 4 to 16,000 concurrent

voice calls. For scalability and redundancy, CUBE offers the following functionality:

The ISR-G2 platforms offer Box-to-box redundancy with media preservation for stable active calls

(CUBE 8.5 in Cisco IOS Release 15.1.2T)

The ASR platforms offer box-to-box or inbox redundancy with media and signaling preservation

(stateful failover) for stable active calls (Release 3.2)

Centralized ApplicationsThe deployment of a Unified CM Session Management Edition enables commonly used applications,

such as conferencing or videoconferencing to connect directly to the session management cluster, which

reduces the overhead of managing multiple trunks to leaf systems.

Unified CM Session Management Edition supports the following applications:

Unity, Unity Connection

Meeting Place, Meeting Place Express

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Before you Begin

SIP and H.323-based video conferencing systems

Third Party voice mail systems

Fax servers

Cisco Unified Mobility

The following applications must connect to the leaf cluster:

Unified Contact Centre, Unified Contact Centre Express

Cisco Unified Presence Server

Attendant Console

Manager Assistant

IP IVR

Cisco Voice Portal

As a general rule, applications that rely only on numbering to establish calls can connect to Unified CM

Session Management Edition. Applications that require additional interfaces or need to track device

state, such as CTI, must connect to the leaf cluster.

If phones are deployed on the Unified CM Session Management Edition cluster, any standard Cisco

Unified CM applications can be deployed to serve these phones.

Before you BeginMindy, reformat and possibly add info to this section to tie all of these loose ends together. Consider a

checklist with these sections linked from the checklist.

Licensing

You can purchase Unified CM Session Management Edition session licenses as part of the hardware and

software purchasing process, or as part of a Cisco Unified CM system upgrade. You can purchase

additional session licenses and install them at any time after the initial Unified CM Session Management

Edition installation.

For information about ordering Unified CM Session Management Edition licenses, refer to the Cisco

Unified Communications Solutions Ordering Guide at the following URL:

http://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/uc_so

l_og_ucl_final.pdf

Unified CM Session Management Edition licenses are based on the number of sessions. The number of

Sessions is the number of Concurrent calls that are supported through the Unified CM Session

Management Edition system.

Interoperability Testing

Although most vendors do conform to standards, differences can and do exist between protocol

implementations from various vendors. As with any standard Cisco Unified CM cluster, Cisco strongly

recommends that you conduct end-to-end system interoperability testing with any unverified third party

unified communications system before deploying the system in a production environment. The

interoperability testing should verify call flows and features from Cisco and third party leaf systems
http://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/us_sol_og_ucl_final.pdfhttp://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/us_sol_og_ucl_final.pdfhttp://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/us_sol_og_ucl_final.pdfhttp://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/us_sol_og_ucl_final.pdf

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Before you Begin

through the Cisco Unified CM cluster. To learn which third party unified communications systems have

been tested by the Cisco Interoperability team, go to http://www.cisco.com/go/interoperability and

select the link for Cisco Unified Communications ManagerSession Management Edition.

CUBE has also been tested for interoperability with different service providers for their SIP trunk

offerings, as well as connectivity to different IP-PBXs. The detailed list and configuration guides are

available at http://www.cisco.com/go/interoperability .Make sure that you have a discussion about system testing with your Cisco account team. Cisco can

provide a generic system test plan if you require it and Cisco advanced services can also be engaged to

conduct onsite interoperability testing.

Installation Considerations

Unified CM Session Management Edition uses exactly the same software as Cisco Unified CM and

supports a subset of Cisco Unified CM hardware platforms. The installation of an Unified CM Session

Management Edition cluster is exactly the same as a Cisco Unified CM cluster.

Unified CM Session Management Edition Design Considerations

Unified CM Session Management Edition cluster designs are a variation on distributed call processing

designs for Cisco Unified CM clusters, as described in the Cisco Unified Communications System 8.x

SRND document. The key difference between Cisco Unified CM and Unified CM Session Management

Edition cluster designs is that, with a Unified CM Session Management Edition design the focus is on

trunks rather than endpoints, such as IP Phones. Use the design guidelines that are provided in the Cisco

Unified Communications System 8.x SRND document for aspects of Unified CM Session Management

Edition cluster design that are not covered in this document. You can access the Cisco Unified

Communications System 8.x SRND document at the following URL:

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_implementation_design_gu

ides_list.htmlAll Unified CM Session Management Edition deployments use either one, or a combination of Cisco

Unified CM trunks to interconnect leaf Cisco Unified CM systems, IP PSTN connections and Cisco

Unified CM applications. The following chapter discusses Cisco Unified CM trunks in detail.
http://www.cisco.com/go/interoperabilityhttp://www.cisco.com/go/interoperabilityhttp://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_implementation_design_guides_list.htmlhttp://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_implementation_design_guides_list.htmlhttp://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_implementation_design_guides_list.htmlhttp://www.cisco.com/go/interoperabilityhttp://www.cisco.com/go/interoperability

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2Cisco Unified Communications Manager Trunks

A trunk is a communications channel on Cisco Unified Communications Manager (Cisco Unified CM)

that enables Cisco Unified CM to connect to other servers. Using one or more trunks, Cisco Unified CM

can receive or place voice, video, and encrypted calls, exchange real-time event information, and

communicate in other ways with call control servers and other external servers.

Trunks are an integral and a crucial part of a Cisco Unified Communications deployment, hence it isimportant to understand the types of trunks available, their capabilities, and design and deployment

considerations such as resiliency, capacity, load balancing, and so forth.

There are two basic types of trunks that can be configured in Cisco Unified CM:

SIP and H.323 trunks, both of which can be used for external communications

Intercluster trunks (ICTs)

This chapter describes the general capabilities and functions of these trunks:

This chapter discusses the following topics:

A Comparison of SIP and H.323 Trunks, page 2-1

SIP Trunks Overview, page 2-4

H.323 Trunks Overview, page 2-34

General SIP and H.323 Trunk Design Considerations, page 2-53

Other Design Considerations for Unified CM Session Management Edition Deployments

A Comparison of SIP and H.323 TrunksCisco Unified CM trunk connections support both SIP and H.323. In many cases, the decision to use SIP

or H.323 is driven by the unique feature(s) offered by each protocol. There are also a number of external

factors that can affect the choice of trunk protocol, such as customer preference or the protocol's maturity

and degree of interoperability offered between various vendors' products.

For trunk connections between Cisco devices, this decision is relatively straightforward. For trunk

connections to other vendors' products and to service provider networks, it is important to understand

which features are required by the customer and the extent of interoperability between any two vendors'

products.

Table 2-1 compares some of the features offered over SIP and H.323 trunks between Cisco Unified CM

clusters.

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Table 2-1 Comparison of SIP and H.323 Features on Cisco Unified Communications ManagerTrunks

Feature SIP QSIG over SIP H.323 QSIG over H.323

Calling Line (Number) Identification Presentation Yes Yes Yes Yes

Calling Line (Number) Identification Restriction Yes Yes Yes Yes

Calling Name Identification Presentation Yes Yes Yes Yes

Calling Name Identification Restriction Yes Yes Yes Yes

Connected Line (Number) Identification Presentation Yes Yes Yes Yes

Connected Line (Number) Identification Restriction Yes Yes Yes Yes

Connected Name Identification Presentation Yes Yes Yes Yes

Connected Name Identification Restriction Yes Yes Yes Yes

Alerting Name Yes Yes No Yes

Call Transfer (Blind/Attended) Yes/Yes Yes/Yes Yes/Yes Yes/Yes

Call Forward All Yes Yes Yes Yes

Call Forward Busy Yes Yes Yes Yes

Call Forward No Reply Yes Yes Yes Yes

Call Completion to Busy Subscriber No Yes No Yes

Call Completion No Reply No Yes No Yes

Subscribe/Notify, Publish Presence Yes Yes No No

Message Waiting Indication (MWI: lamp ON,

lamp OFF)

Yes Yes No Yes

Path Replacement No Yes No Yes

Call Hold/Resume Yes Yes Yes Yes

Music On Hold (unicast and multicast) Yes Yes Yes YesDTMF-relay RFC 2833,

KPML (OOB),

Unsolicited

Notify (OOB)

RFC 2833,

KPML (OOB),

Unsolicited

Notify (OOB)

H.245 Out Of

Band (OOB)1H.245 Out Of

Band (OOB)1

SIP Early Offer Yes MTP

may be

required

Yes MTP

may be

required

N/A N/A

SIP Delayed Offer Yes Yes N/A N/A

H.323 Fast Start N/A N/A Yes MTP

always

required forOutbound Fast

Start

Yes MTP

always required

for OutboundFast Start

H.323 Slow Start N/A N/A Yes Yes

Audio codecs G.711, G.722,

G.723, G.729,

iLBC, AAC,

iSAC

G.711, G.722,

G.723, G.729,

iLBC, AAC,

iSAC

G.711, G.722,

G.723, G.729

G.711, G.722,

G.723, G.729

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A Comparison of SIP and H.323 Trunks

Codecs with MTP All codecs

supported

when EarlyOffer support

for voice and

video calls

(insert MTP if

needed) is

checked

G.711, G.729

when MTP

Required is

checked

All codecs

supported

when EarlyOffer support

for voice and

video calls

(insert MTP if

needed) is

checked

G.711, G.729

when MTP

Required is

checked

G.711, G.723,

G.729

G.711, G.723,

G.729

Video Yes Yes Yes Yes

Video codecs H.261, H.263,H.263+,

H.264 AVC

H.261, H.263,H.263+,

H.264 AVC

H.261, H.263,H.263+,

H.264 AVC

H.261, H.263,H.263+,

H.264 AVC

T.38 Fax Yes Yes Yes Yes

Signaling Authentication Digest, TLS Digest, TLS No No

Signaling Encryption TLS TSL No No

Media Encryption (audio) SRTP SRTP SRTP SRTP

RSVP-based QoS and call admission control Yes Yes No No

Support for + character Yes Yes No No

Inbound Calls Called Party: Significant Digits,

Prefix-Digits

Yes Yes Yes Yes

Incoming Calling Party Settings: Strip Digits,

Prefix-Digits based on Number Type

SIP does not

support

Number Type -

"Unknown"

used for all

calls

SIP does not

support

Number Type -

"Unknown"

used for all

calls

Cisco Unified

CM, Unknown,

National,

International,

Subscriber

Cisco Unified

CM, Unknown,

National,

International,

Subscriber

Incoming Called Party Settings: Strip Digits,

Prefix-Digits based on Number Type

N/A N/A Cisco Unified

CM, Unknown,

National,

International,

Subscriber

Cisco Unified

CM, Unknown,

National,

International,

Subscriber

Connected Party Transformation Yes Yes No No

Outbound Calling Party Transformations Yes Yes Yes Yes

Outbound Called Party Transformations Yes Yes Yes Yes

Table 2-1 Comparison of SIP and H.323 Features on Cisco Unified Communications ManagerTrunks (continued)


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SIP Trunks OverviewSIP trunks on Cisco Unified CM/Unified CM Session Management Edition can be used for two different

purposes:

Intra-enterprise SIP trunks provide connectivity to other SIP devices such as gateways, Unified CM

Session Management Edition, SIP proxies, Unified Communications applications, and other Cisco

Unified CM clusters within the enterprise network.

Service provider SIP trunks provide offnet IP PSTN connectivity to a service provider network.

Today, SIP is arguably the most commonly chosen protocol when connecting to service providers and

Unified Communications applications. Cisco Unified CM 8.5 and later releases provide the following

SIP trunk and call routing enhancements:

Run on all Cisco Unified CM nodes

Up to 16 destination IP addresses per trunk

SIP OPTIONS ping keepalives

SIP Early Offer support for voice and video calls (insert MTP if needed)

QSIG over SIP

SIP trunk normalization and transparency

Run route lists on all Cisco Unified CM nodes

The SIP trunk features available in the 8.5 release make SIP the preferred choice for new and existing

trunk connections. The QSIG over SIP feature provides parity with H.323 intercluster trunks and canalso be used to provide QSIG over SIP trunk connections to Cisco IOS gateways (and on to QSIG-based

TDM PBXs). The ability to run on all Cisco Unified CM nodes and to handle up to 16 destination IP

addresses improves outbound call distribution from Cisco Unified CM clusters and reduces the number

of SIP trunks required between clusters and devices. SIP OPTIONS ping provides dynamic reachability

detection for SIP trunk destinations, rather than per-call reachability determination. SIP Early Offer

support for voice and video calls (insert MTP if needed) can reduce or eliminate the need to use MTPs

and allows voice, video, and encrypted calls to be made over SIP Early Offer trunks.

Outbound Calling/Called Party Number Type Setting SIP does not

support

Number Type

SIP does not

support

Number Type

Cisco Unified

CM, Unknown,

National,International,

Subscriber

Cisco Unified

CM, Unknown,

National,International,

Subscriber

Outbound Called/Called Party Numbering Plan

Setting

SIP does not

support

Number Plan

SIP does not

support

Number Plan

Cisco Unified

CM, ISDN,

National

Standard,

Private,

Unknown

Cisco Unified

CM, ISDN,

National

Standard,

Private,

Unknown

Trunk destination State detection mechanism OPTIONS Ping OPTIONS Ping Per call attempt Per call attempt

1. H.323 trunks support signaling of RFC 2833 for certain connection types.

Table 2-1 Comparison of SIP and H.323 Features on Cisco Unified Communications ManagerTrunks (continued)


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SIP trunk normalization and transparency improve native Cisco Unified CM interoperability with and

between third-party unified communications systems. Normalization allows inbound and outbound SIP

messages and SDP information to be modified on a per-SIP-trunk basis. Transparency allows Cisco

Unified CM to pass SIP headers, parameters, and content bodies from one SIP trunk call leg to another,

even if Cisco Unified CM does not understand or support the parts of the message that are being passed

through.

These features are discussed in detail later in this section.

For the complete list of new enhancements for SIP trunks, refer to the New and Changed for Cisco

Unified Communications Manager 8.5(1) document available at

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html

General Deployment Considerations

Cisco Unified CM SIP trunks offer a greater set of features in comparison with H.323 intercluster trunks,

thus making SIP the protocol of choice for intercluster trunk connections (although H.323 Annex M1

may still be preferred for intercluster trunk connections to Cisco Unified CM clusters using earlier

software versions). Also, given the wide support of SIP in the industry, SIP trunks are usually a goodchoice for connectivity to third-party applications and service providers.

SIP Trunk Features and Operation

This section explains how Cisco Unified CM SIP trunks operate and describes several key SIP trunk

features that should be taken into account when designing and deploying Cisco Unified CM SIP trunks.

SIP Trunks Can Run on All Active Unified CM Nodes

When the Run on all Active Unified CM Nodes option is checked on a SIP trunk, Cisco Unified CM

creates an instance of the SIP trunk daemon on every call processing subscriber within the cluster, thusallowing SIP trunk calls to be made or received on any call processing subscriber. (Prior to this feature,

up to three nodes could be selected per trunk by using Cisco Unified CM Groups.) With Run on all

Active Unified CM Nodes enabled, outbound SIP trunk calls originate from the same node on which

the inbound call (for example, from a phone or trunk) is received. As with all Cisco Unified CM SIP

trunks, the SIP daemons associated with the trunk will accept inbound calls only from end systems with

IP addresses that are defined in the trunk's destination address fields. Running SIP trunks on all nodes

is recommended where the SIP trunk is required to process a large number of calls so that outbound and

inbound call distribution can be evenly spread across all call processing subscribers within a cluster.

Also, when multiple SIP trunks to the same destination(s) are using the same subscriber, a unique

incoming and destination port number must be defined per trunk to allow each trunk to be identified

uniquely.

Up to 16 SIP Trunk Destination IP Addresses

SIP trunks can be configured with up to 16 destination IP addresses, 16 fully qualified domain names,

or a single DNS SRV entry. Support for additional destination IP addresses reduces the need to create

multiple trunks associated with route lists and route groups for call distribution between two Unified

Communications systems, thus simplifying Cisco Unified CM trunk design. (See Figure 2-1.) This

feature can be used in conjunction with the Run on all Active Unified CM Nodes feature or with a SIP

trunk that uses standard Cisco Unified CM Groups to create a SIP daemon on up to three nodes within
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.htmlhttp://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html

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SIP Trunks Overview

the cluster. Bear in mind, however, that the SIP daemons associated with a Cisco Unified CM SIP trunk

will accept inbound calls only from end systems with IP addresses that are defined in the trunk's

destination address fields.

Figure 2-1 SIP Trunks with Multiple Destination IP Addresses Running on All Active Nodes

SIP OPTIONS Ping

The SIP OPTIONS Ping feature can be enabled on the SIP Profile associated with a SIP trunk to

dynamically track the state of the trunk's destination(s). When this feature is enabled, each node running

the trunk's SIP daemon will periodically send an OPTIONS Request to each of the trunk's destination IP

addresses to determine its reachability and will send calls only to reachable nodes. A destination address

is considered to be "out of service" if it fails to respond to an OPTIONS Request, if it sends a Service

Unavailable (503) response or Request Timeout (408) response, or if a TCP connection cannot be

established. The trunk state is considered to be "in service" when at least one node receives a response(other than a 408 or 503) from a least one destination address. SIP trunk nodes can send OPTIONS

Requests to the trunk's configured destination IP addresses or to the resolved IP addresses of the trunk's

DNS SRV entry. Enabling SIP OPTIONS Ping is recommended for all SIP trunks because it allows

Cisco Unified CM to dynamically track trunk state rather than determining trunk state on a per-call and

timeout basis.

SIP Early Offer Support over Cisco Unified CM SIP Trunks

SIP negotiates media exchange by means of the Session Description Protocol (SDP), where one side

offers a set of capabilities to which the other side answers, thus converging on a set of media

characteristics. SIP allows the initial offer to be sent either by the caller in the initial INVITE message

(Early Offer) or, if the caller chooses not to, the called party can send the initial offer in the first reliableresponse (Delayed Offer).

By default, Cisco Unified CM SIP trunks send the INVITE without an initial offer (Delayed Offer).

Cisco Unified CM has two configurable options to enable a SIP trunk to send the offer in the INVITE

(Early Offer):

Media Termination Point Required, page 2-7

Early Offer Support for Voice and Video Calls (Insert MTP If Needed), page 2-7

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Media Termination Point Required

Enabling the Media Termination Point Required option on the SIP trunk assigns an MTP from the

trunk's media resources group (MRG) to every outbound call. (See Figure 2-2.) This statically assigned

MTP supports only the G.711 or G.729 codecs, thus limiting media to voice calls only.

Figure 2-2 SIP Early Offer with Media Termination Point Required

Early Offer Support for Voice and Video Calls (Insert MTP If Needed)

Enabling Early Offer support for voice and video calls (insert MTP if needed) on the SIP Profile

associated with the SIP trunk inserts an MTP only if the calling device cannot provide Cisco Unified CM

with the media characteristics required to create the Early Offer. In general, Early Offer support for

voice and video calls (insert MTP if needed) is recommended because this configuration option

reduces MTP usage (see Figure 2-3). Calls from older SCCP-based phones registered to Cisco Unified

CM over SIP Early Offer trunks configured with this option will use an MTP to create the Offer SDP,

and these calls support voice, video, and encrypted media (see Endpoint Features Summary,page 19-50). Inbound calls to Cisco Unified CM from SIP Delayed Offer trunks or H.323 Slow Start

trunks that are extended over an outbound SIP Early Offer trunk will use an MTP to create the Offer

SDP; however, these calls support audio only in the initial call set up but can be escalated mid-call to

support video and SRTP if the call media is renegotiated (for example, after hold/resume). For guidance

on when to use Early Offer support for voice and video calls (insert MTP if needed) , see Design

Considerations for SIP Trunks, page 2-25.

Note MTP resources are not required for incoming INVITE messages, whether or not they contain an initial

offer SDP.

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Figure 2-3 Early Offer Support for Voice and Video Calls

Cisco Unified CM does not need to insert an MTP to create an outbound Early Offer call over a SIP trunk

if the inbound call to Cisco Unified CM is received by any of the following means:

On a SIP trunk using Early Offer

On an H.323 trunk using Fast Start

On an MGCP trunk

From a SIP-based IP phone registered to Cisco Unified CM

From newer SCCP-based Cisco Unified IP Phone models registered to Cisco Unified CM (see the

Endpoint Features Summary, page 19-50, for details)For the above devices, Cisco Unified CM uses the media capabilities of the endpoint and applies the

codec filtering rules based on the region-pair of the calling device and outgoing SIP trunk to create the

offer SDP for the outbound SIP trunk. In most cases, the offer SDP will have the IP address and port

number of the endpoint initiating the call. This is assuming that Cisco Unified CM does not have to insert

an MTP for other reasons such as a DTMF mismatch, TRP requirements, or a transcoder requirement

when there is no common codec between the regions of the calling device and the SIP trunk.

When Early Offer support for voice and video calls (insert MTP if needed) is configured on a trunk's

SIP Profile, calls from older SCCP-based phones (see Endpoint Features Summary, page 19-50), SIP

Delayed Offer trunks, and H.323 Slow Start trunks will cause Cisco Unified CM to allocate an MTP if

an MTP or transcoder has not already been allocated for that call for another reason. The MTP is used

to generate an offer SDP with a valid media port and IP address. The MTP will be allocated from the

media resources associated with the calling device rather than from the outbound SIP trunk's mediaresources. (This prevents the media path from being anchored to the outbound SIP trunk's MTP). If the

MTP cannot be allocated from the calling device's media resource group list (MRGL), then the MTP

allocation is attempted from the SIP trunk's MRGL.

For calls from older SCCP phones (see Endpoint Features Summary, page 19-50) registered to Cisco

Unified CM, some of the media capabilities of the calling device (for example, supported voice codecs,

video codecs, and encryption keys if supported) are available for media exchange through the Session

Description Protocol (SDP). Cisco Unified CM will create a superset of the endpoint and MTP codec

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capabilities and apply the codec filtering based on the applicable region-pair settings. The outbound

Offer SDP will use the MTP's IP address and port number and can support voice, video, and encrypted

media. Note that the MTP should be configured to support the pass-through codec.

When Cisco Unified CM receives an inbound call on an H.323 Slow Start or SIP Delayed Offer trunk,

the media capabilities of the calling device are not available when the call is initiated. In this case, Cisco

Unified CM must insert an MTP and will use its IP address and UDP port number to advertise allsupported audio codecs (after region pair filtering) in the Offer SDP of the initial INVITE sent over the

outbound SIP trunk. When the Answer SDP is received on the SIP trunk, if it contains a codec that is

supported by the calling endpoint, then no additional offer-answer transaction is needed. In case of codec

mismatch, Cisco Unified CM can either insert a transcoder to address the mismatch or send a reINVITE

or UPDATE to trigger media negotiation. Calls from H.323 Slow Start or SIP Delayed Offer trunks

support audio only in the initial call setup, but they can be escalated mid-call to support video and SRTP

if the call media is renegotiated (for example, after Hold/Resume).

QSIG over SIP Trunks

Cisco Unified CM can encapsulate QSIG content in SIP messages, thus allowing features such as Call

Back, MWI, and Path Replacement to be invoked over SIP QSIG intercluster trunks and over SIP QSIG

trunks to Cisco IOS gateways. (See Figure 2-4.) QSIG over SIP trunks provides parity with the QSIG

feature set on H.323 Annex M1 intercluster trunks and MGCP QSIG trunks. (ISO and ECMA variants

of QSIG are supported on a per-trunk basis.)

Figure 2-4 QSIG over SIP Trunks

SIP Trunk Message Normalization and Transparency

Normalization and transparency provide powerful script-based functionality for SIP trunks that can be

used to transparently forward and/or modify SIP messages and message body contents as they traverse

Cisco Unified CM. Normalization and transparency scripts are designed to address SIP interoperability

issues, allowing Cisco Unified CM to interoperate with SIP-based third-party PBXs, applications, and

IP PSTN services.

SIP Trunk Normalization

Normalization allows incoming and outgoing SIP messages to be modified on their way through Cisco

Unified CM. Normalization applies to all calls that traverse a SIP trunk with an associated script,

regardless of what protocol is being used for the other endpoint involved in the call. For example, a SIP

trunk normalization script can operate on a call from a SIP line device to a SIP trunk, from an

SCCP-based device to a SIP trunk, from MGCP to SIP trunk, from H.323 to SIP trunk, and so forth. (See

Figure 2-5.) Normalization does not require end-to-end SIP.

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Figure 2-5 SIP Trunk Normalization

SIP Trunk Transparency

Transparency allows Cisco Unified CM to pass SIP headers, parameters, and content bodies from one

SIP trunk call leg to another, even if Cisco Unified CM does not understand or support the parts of the

message that are being passed through. Transparency (or transparent pass-through) is applicable only

when the call through Cisco Unified CM is from SIP trunk to SIP trunk, as illustrated in Figure 2-6.

Figure 2-6 SIP Trunk Transparency

Normalization and transparency scripts use Lua, a powerful, fast, lightweight, embeddable scriptinglanguage to modify SIP messages and SDP body content on SIP trunks. (For more information on Lua,

refer to the documentation available at http://lua-users.org/wiki/LuaOrgGuide .)

Cisco has created a library of Lua-based SIP Message APIs that allow specified information in the SIP

message and SDP body to be retrieved, modified, replaced, removed, passed through, ignored, appended

to, transformed, and so on. The underlying Lua language allows retrieved information to be stored as

variables and operated on using a series of operations such as: If, elseif, while, do, , =, and so forth.

The scripting approach naturally supports multiple variables and state-specific contexts for making

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script decisions. The combination of Cisco's SIP Message Library APIs and the functionality underlying

the Lua language creates a very powerful scripting environment that allows almost any SIP message

and/or its SDP body content to be modified.

For inbound messages on a SIP trunk, normalization and transparency script processing occurs

immediately after receiving the message from the network. For outbound messages, script processing

occurs immediately before sending the message to the network.Within a Lua script, callback functions (also known as message handlers) are used to request message

types of interest. The Cisco Lua environment constructs the name of the message handler based on the

message direction and method for requests (for example, inbound_INVITE) and based on the message

direction, response code, and method (from the CSeq header) for responses (for example,

outbound_180_INVITE). A message object (for example, msg) is passed to the message handler, thereby

allowing the script to modify the message (for example, inbound_INVITE(msg)).

Callback Function (message Handler) examples:

The Lua script then uses APIs defined in the Cisco SIP Message library to access and manipulate

message parameters. For example:

getHeader(header-name) returns header-value or ""

getHeaderValues(header-name) returns a table of header values

addHeaderValueParameter(header-name,parameter-name, [parameter-value])

getUri(header-name) retrieves the URI from the specified header

block() blocks the specified SIP message

applyNumberMask(header-name, mask) retrieves the specified header and applies the specified

number mask to the URI getSdp() returns the SDP content

sdp:getLine(start of line, line contains) returns line in SDP that starts with "start of line" and also

has string "line contains"

sdp:modifyLine(start of line, line contains,new-line) finds the in SDP that starts with "start of

line", the line matching "line contains" is replaced with the new-line parameter

The following examples illustrate the use of SIP Message API scripts.

Example 2-1 SIP Message API getRequestLine

getRequestLine() returns the method, request-uri, and version.

This method returns three values:

The method name

The request-uri

The protocol version

inbound_INVITE() outbound_INVITE()

inbound_UPDATE() outbound_SUBSCRIBE()

inbound_3XX_INVITE() outbound_180_INVITE()

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Example script:

Line 1 initializes the set of callback functions to an empty value. This set of callback functions, named

M, is essentially a Lua table.

Lines 2 to 4 define a message handler. This callback function is executed when an outbound INVITE is

sent from Cisco Unified CM. The script then gets the method, request-uri, and version from the request

line and stores these values.

The script can define multiple message handlers. The name of the message handler dictates which

message handler is invoked (if any) for a given SIP message.

The last line returns the set of callbacks. This line is absolutely required.Message:

INVITE sip:[email protected] SIP/2.0

Output and result:

method == "INVITE"

ruri == "sip:[email protected]"

version == "SIP/2.0"

Example 2-2 A script that simply removes the "Cisco-Guid" header in an outbound INVITE

Line 1 initializes the set of callback functions to an empty value. This set of callback functions, named

M, is essentially a Lua Table.

Lines 2 to 4 define a message handler. This callback function is executed when an outbound INVITE is

sent from Cisco Unified CM. The script can define multiple message handlers. The name of the message

handler dictates which message handler is invoked (if any) for a given SIP message.

The last line returns the set of callbacks. This line is absolutely required.

Message:


.

P-Asserted-Identity: "1234"

Cisco-Guid: 1234-4567-1234

Session-Expires: 1800

Line 1 M = {}

Line 2 function M.outbound_INVITE(message)

Line 3 local method, ruri, ver = message:getRequestLine()

Line 4 end

Line 5 return M

Line 1 M = {}

Line 2 function M.outbound_INVITE(message)

Line 3 message:removeHeader("Cisco-Guid")

Line 4 end

Line 5 return M

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Output and results:


.

P-Asserted-Identity: "1234"

For more information on SIP trunk normalization and transparency scripts, refer to theDeveloper Guide

for SIP Transparency and Normalization, available at

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/sip_tn/8_5_1/sip_t_n.html

Route Lists Run on All Active Unified CM Nodes

Although this is not specifically a SIP trunk feature, running route lists on all nodes provides benefits

for trunks in route lists and route groups. Running route lists on all nodes improves outbound call

distribution by using the "route local" rule to avoid unnecessary intra-cluster traffic.

For route lists, the route local rule operates as follows:

For outbound calls that use route lists (and associated route groups and trunks), when a call from a

registered phone or inbound trunk arrives at the node with the route list instance, Cisco Unified CM

checks to see if an instance of the selected outbound trunk exists on the same node as the route list.If so, Cisco Unified CM will use this

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