MITEL – SIP CoE

Technical
Configuration Notes

Configure the Mitel 3300 for use with

Cisco Call Manager
SIP CoE 09-4940-00068
 NOTICE

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Mitel Technical Configuration Notes – Configure the Mitel 3300 for use with Cisco Call Manager

March 2010, 09-4940-00068_3

®,™ Trademark of Mitel Networks Corporation

© Copyright 2009, Mitel Networks Corporation
All rights reserved

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 Table of Contents

OVERVIEW ............................................................................................................... 1
Interop History....................................................................................................................1
Interop Status ....................................................................................................................1
Software & Hardware Setup...............................................................................................1

CONFIGURATION NOTES ....................................................................................... 2

3300 Configuration Notes ..................................................................................................2
Network Requirements.................................................................................................................... 2
Assumptions for the 3300 Programming......................................................................................... 2
Licensing and Option Selection – SIP Licensing ............................................................................ 3
Class of Service Assignment .......................................................................................................... 4
SIP Peer Profile............................................................................................................................... 5

CISCO CALL MANAGER CONFIGURATION NOTES ............................................. 7

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Overview
This document provides a reference to Mitel Authorized Solutions providers for configuring
the Mitel 3300 to connect to a Cisco Call Manager The different devices can be configured in
various configurations depending on your VoIP solution. This document covers a basic setup
with required option setup.

Interop History

Version Date Reason

1 June , 2009 Field Assessed Interop with Mitel 3300 9.0 and Cisco Call Manager
2 November, 2009 Documentation update to include Cisco Delayed Offer and Early
Offer configuration
3 March, 2010 Documentation update

Interop Status
The Interop of Cisco Call Manager has been given a Certification status. This service
provider or trunking device will be included in the SIP CoE Reference Guide. The status
Cisco Call Manager achieved is:

For informational purposes only, field-assessed means that the Cisco

Call Manager has been tested and/or used to some degree by
someone successfully, though details may or may not be available.
Mitel product support does NOT apply to field-assessed interops.

Software & Hardware Setup

This was the test setup to generate a basic SIP call between Cisco Call Manager and the
3300 .

Manufacturer Variant Software Version

Mitel 3300 – Mxe Platform 9.0.2.18

Cisco Cisco Call Manager 5.1.2.3122-1

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Configuration Notes
This section is a description of how the SIP Interop was configured. These notes should give
a guideline how a device can be configured in a customer environment and how the Cisco
Call Manager and 3300 programming was configured in our test environment.

Disclaimer: Although Mitel has attempted to setup the interop testing facility as closely
as possible to a customer premise environment, implementation setup could be
different onsite. YOU MUST EXERCISE YOUR OWN DUE DILIGENCE IN REVIEWING,
planning, implementing, and testing a customer configuration.

3300 Configuration Notes

The following steps show how to program a 3300 to interconnect with the Cisco Call
Manager.

Network Requirements
• There must be adequate bandwidth to support the voice over IP. As a guide, the Ethernet
bandwidth is approx 85 Kb/s per G.711 voice session and 29 Kb/s per G.729 voice
session (assumes 20ms packetization). As an example, for 20 simultaneous SIP
sessions, the Ethernet bandwidth consumption will be approx 1.7 Mb/s for G.711 and
0.6Mb/s. Almost all Enterprise LAN networks can support this level of traffic without any
special engineering. Please refer to the 3300 Engineering guidelines for further
information.
• For high quality voice, the network connectivity must support a voice-quality grade of
service (packet loss <1%, jitter < 30ms, one-way delay < 80ms).

Assumptions for the 3300 Programming

• The SIP signaling connection uses UDP on Port 5060.

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Licensing and Option Selection – SIP Licensing

Ensure that the 3300 is equipped with enough SIP trunking licenses for the connection to the
Cisco Call Manager This can be verified within the License and Option Selection form.

Enter the total number of licenses in the SIP Trunk Licences field. This is the maximum
number of SIP trunk sessions that can be configured in the 3300 to be used with all service
providers, applications and SIP trunking devices.

Figure 1 – License and Option Selection

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Class of Service Assignment

The Class of Service Options Assignment form is used to create or edit a Class of Service
and specify its options. Classes of Service, identified by Class of Service numbers, are
referenced in the Trunk Service Assignment form for SIP trunks.

Many different options may be required for your site deployment, but ensure that “Public
Network Access via DPNSS” Class of Service Option is configured for all devices that make
outgoing calls through the SIP trunks in the 3300.

• Public Network Access via DPNSS set to Yes

• Campon Tone Security/FAX Machine set to Yes
• Busy Override Security set to Yes

Figure 2 – Class of Service

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SIP Peer Profile

The recommended connectivity via SIP Trunking does not require additional physical
interfaces. IP/Ethernet connectivity is part of the base 3300 Platform. The SIP Peer Profile
should be configured with the following options:

NOTE: Ensure the remaining SIP Peer profile policy options are similar the screen
capture below.

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Figure 3 – SIP Peer Profile Assignment

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Cisco Call Manager Configuration Notes

Device Information
Product: SIP Trunk
Device Protocol: SIP

CTS-Ops-A
Device Name
Description SIP Trunk to Mitel phone system

DP_G711-All
Device Pool

OnNet
Call Classification
Media Resource Group List < None >

Main
Location
AAR Group < None >

None
Packet Capture Mode
Packet Capture Duration 0

Media Termination Point Required

Retry Video Call as Audio

Transmit UTF-8 for Calling Party Name

Unattended Port
Multilevel Precedence and Preemption (MLPP) Information
MLPP Domain < None > 0 0

Call Routing Information Inbound Calls

All
Significant Digits

Default
Connected Line ID Presentation

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Default
Connected Name Presentation
Calling Search Space CS_Gatew ay_Internal

AAR Calling Search Space < None >

Prefix DN

Redirecting Diversion Header Delivery - Inbound

Outbound Calls
Originator
Calling Party Selection

Default
Calling Line ID Presentation

Default
Calling Name Presentation
Caller ID DN

Caller Name

Redirecting Diversion Header Delivery - Outbound

SIP Information
10.106.20.10
Destination Address

Destination Address is an SRV

5060
Destination Port

711ulaw
MTP Preferred Originating Codec

Standard Presence group

Presence Group

Non Secure SIP Trunk Profile

SIP Trunk Security Profile
Rerouting Calling Search Space < None >

Out-Of-Dialog Refer Calling Search Space < None >

SUBSCRIBE Calling Search Space < None >

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Standard SIP Profile

SIP Profile

No Preference
DTMF Signaling Method

Device Information
Product: SIP Trunk
Device Protocol: SIP

CTS-Ops-B
Device Name
Description SIP Trunk to Mitel 3300

DP_G711-All
Device Pool

Use System Default

Call Classification
Media Resource Group List < None >

Hub_None
Location
AAR Group < None >

None
Packet Capture Mode
Packet Capture Duration 0

Media Termination Point Required

Retry Video Call as Audio

Transmit UTF-8 for Calling Party Name

Unattended Port
Multilevel Precedence and Preemption (MLPP) Information
MLPP Domain < None > 0 0

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Call Routing Information Inbound Calls

All
Significant Digits

Default
Connected Line ID Presentation

Default
Connected Name Presentation
Calling Search Space CS_Gatew ay_Internal

AAR Calling Search Space < None >

Prefix DN

Redirecting Diversion Header Delivery - Inbound

Outbound Calls
Originator
Calling Party Selection

Default
Calling Line ID Presentation

Default
Calling Name Presentation
Caller ID DN

Caller Name

Redirecting Diversion Header Delivery - Outbound

SIP Information
10.106.20.15
Destination Address

Destination Address is an SRV

5060
Destination Port

711ulaw
MTP Preferred Originating Codec

Standard Presence group

Presence Group

Non Secure SIP Trunk Profile

SIP Trunk Security Profile

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Rerouting Calling Search Space < None >

Out-Of-Dialog Refer Calling Search Space < None >

SUBSCRIBE Calling Search Space < None >

Standard SIP Profile

SIP Profile

No Preference
DTMF Signaling Method

SIP Profile Information

Standard SIP Profile
Name
Description Default SIP Profile

101
Default MTP Telephony Event Payload Type

Redirect by Application

Disable Early Media on 180

Parameters used in Phone
180
Timer Invite Expires (seconds)

5
Timer Register Delta (seconds)

3600
Timer Register Expires (seconds)

500
Timer T1 (msec)

4000
Timer T2 (msec)

6
Retry INVITE

10
Retry Non-INVITE

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16384
Start Media Port

32766
Stop Media Port

x-cisco-serviceuri-pickup
Call Pickup URI

x-cisco-serviceuri-opickup
Call Pickup Group Other URI

x-cisco-serviceuri-gpickup
Call Pickup Group URI

x-cisco-serviceuri-meetme
Meet Me Service URI

None
User Info

Nominal
DTMF DB Level

Off
Call Hold Ring Back

Off
Anonymous Call Block

Off
Caller ID Blocking

User
Do Not Disturb Control

Disabled
Telnet Level for 7940 and 7960

120
Timer Keep Alive Expires (seconds)

120
Timer Subscribe Expires (seconds)

5
Timer Subscribe Delta (seconds)

70
Maximum Redirections
Off Hook To First Digit Timer
15000
(microseconds)

x-cisco-serviceuri-cfw dall
Call Forward URI

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x-cisco-serviceuri-abbrdial
Abbreviated Dial URI

Conference Join Enabled

RFC 2543 Hold

Semi Attended Transfer

Enable VAD

Stutter Message Waiting

Call Stats

SIP Trunk Security Profile Information

Non Secure SIP Trunk Profile
Name
Description Non Secure SIP Trunk Profile authenticated by null St

Device Security Mode Non Secure

TCP+UDP
Incoming Transport Type
Outgoing Transport Type TCP

Enable Digest Authentication

600
Nonce Validity Time (mins)
X.509 Subject Name

5060
Incoming Port

Enable Application Level Authorization

Accept Presence Subscription

Accept Out-of-Dialog REFER

Accept Unsolicited Notification

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Accept Replaces Header

SIP Delayed Offer and Early Offer

Cisco Unified CM uses the SIP Offer/Answer model for establishing SIP sessions, as defined
in RFC 3264. In this context, an Offer is contained in the Session Description Protocol (SDP)
fields sent in the body of a SIP message. The Offer typically defines the media characteristics
supported by the device (media streams, codecs, directional attributes, IP address, and ports
to use). The device receiving the Offer sends an Answer in the SDP fields of its SIP
response, with its corresponding matching media streams and codec, whether accepted or
not, and the IP address and port on which it wants to receive the media streams. Unified CM
uses this Offer/Answer model to establish SIP sessions as defined in the key SIP standard,
RFC 3261.
RFC 3261 defines two ways that SDP messages can be sent in the Offer and Answer. These
methods are commonly known as Delayed Offer and Early Offer, and support for both
methods by User Agent Client/Servers is a mandatory requirement of the specification. In the
simplest terms, an initial SIP Invite sent with SDP in the message body defines an Early
Offer, whereas an initial SIP Invite without SDP in the message body defines a Delayed
Offer.
In an Early Offer, the session initiator (calling device) sends its capabilities (for example,
codecs supported) in the SDP contained in the initial Invite (thus allowing the called device
to choose its preferred codec for the session). In a Delayed Offer, the session initiator does
not send its capabilities in the initial Invite but waits for the called device to send its
capabilities first (for example, the list of codecs supported by the called device, thus allowing
the calling device to choose the codec to be used for the session).
Delayed Offer and Early Offer are the two media capabilities exchange options available to
all standards-based SIP switches. Most vendors have a preference for either Delayed Offer or
Early Offer, each of which has its own set of benefits and limitations.

It is recommended to enable Early Offer on the Cisco CCM to integrate with the Mitel
3300 ICP.

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