09-4940-00068_3 - cisco call manager
TRANSCRIPT
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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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
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
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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.