09-4940-00068_3 - Cisco Call Manager

MITEL – SIP CoE

Technical Configuration Notes

Configure the Mitel 3300 for use with Cisco Call Manager

SIP CoE 09-4940-00068

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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

Table of Contents

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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

09-4940-00068_3 Cisco Call Manager

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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

Device Name CTS-Ops-A

Description SIP Trunk to Mitel phone system

Device Pool DP_G711-All

Call Classification OnNet

Media Resource Group List < None >

Location Main

AAR Group < None >

Packet Capture Mode None

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

Significant Digits All

Connected Line ID Presentation Default


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Connected Name Presentation Default

Calling Search Space CS_Gatew ay_Internal

AAR Calling Search Space < None >

Prefix DN

Redirecting Diversion Header Delivery - Inbound

Outbound Calls

Calling Party Selection Originator

Calling Line ID Presentation Default

Calling Name Presentation Default

Caller ID DN

Caller Name

Redirecting Diversion Header Delivery - Outbound

SIP Information

Destination Address 10.106.20.10

Destination Address is an SRV

Destination Port 5060

MTP Preferred Originating Codec 711ulaw

Presence Group Standard Presence group

SIP Trunk Security Profile Non Secure SIP Trunk Profile

Rerouting Calling Search Space < None >

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

SUBSCRIBE Calling Search Space < None >


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SIP Profile Standard SIP Profile

DTMF Signaling Method No Preference

Device Information

Product: SIP Trunk

Device Protocol: SIP

Device Name CTS-Ops-B

Description SIP Trunk to Mitel 3300

Device Pool DP_G711-All

Call Classification Use System Default

Media Resource Group List < None >

Location Hub_None

AAR Group < None >

Packet Capture Mode None

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

Significant Digits All

Connected Line ID Presentation Default

Connected Name Presentation Default

Calling Search Space CS_Gatew ay_Internal

AAR Calling Search Space < None >

Prefix DN

Redirecting Diversion Header Delivery - Inbound

Outbound Calls

Calling Party Selection Originator

Calling Line ID Presentation Default

Calling Name Presentation Default

Caller ID DN

Caller Name

Redirecting Diversion Header Delivery - Outbound

SIP Information

Destination Address 10.106.20.15

Destination Address is an SRV

Destination Port 5060

MTP Preferred Originating Codec 711ulaw

Presence Group Standard Presence group

SIP Trunk Security Profile Non Secure SIP Trunk Profile


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Rerouting Calling Search Space < None >

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

SUBSCRIBE Calling Search Space < None >

SIP Profile Standard SIP Profile

DTMF Signaling Method No Preference

SIP Profile Information

Name Standard SIP Profile

Description Default SIP Profile

Default MTP Telephony Event Payload Type 101

Redirect by Application

Disable Early Media on 180

Parameters used in Phone

Timer Invite Expires (seconds) 180

Timer Register Delta (seconds) 5

Timer Register Expires (seconds) 3600

Timer T1 (msec) 500

Timer T2 (msec) 4000

Retry INVITE 6

Retry Non-INVITE 10


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Start Media Port 16384

Stop Media Port 32766

Call Pickup URI x-cisco-serviceuri-pickup

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

Call Pickup Group URI x-cisco-serviceuri-gpickup

Meet Me Service URI x-cisco-serviceuri-meetme

User Info None

DTMF DB Level Nominal

Call Hold Ring Back Off

Anonymous Call Block Off

Caller ID Blocking Off

Do Not Disturb Control User

Telnet Level for 7940 and 7960 Disabled

Timer Keep Alive Expires (seconds) 120

Timer Subscribe Expires (seconds) 120

Timer Subscribe Delta (seconds) 5

Maximum Redirections 70

Off Hook To First Digit Timer

(microseconds) 15000

Call Forward URI x-cisco-serviceuri-cfw dall


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Abbreviated Dial URI x-cisco-serviceuri-abbrdial

Conference Join Enabled

RFC 2543 Hold

Semi Attended Transfer

Enable VAD

Stutter Message Waiting

Call Stats

SIP Trunk Security Profile Information

Name Non Secure SIP Trunk Profile

Description Non Secure SIP Trunk Profile authenticated by null St

Device Security Mode Non Secure

Incoming Transport Type TCP+UDP

Outgoing Transport Type TCP

Enable Digest Authentication

Nonce Validity Time (mins) 600

X.509 Subject Name

Incoming Port 5060

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.

09-4940-00068_3 - Cisco Call Manager

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