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Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Level 3 SIP Trunking
with Avaya Aura® Communication Manager Evolution
Server, Avaya Aura® Session Manager, and Avaya Aura®
Session Border Controller – Issue 1.0
Abstract
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between Level 3 SIP Trunking and an Avaya SIP-enabled enterprise solution. The
Avaya solution consists of Avaya Aura® Session Manager R6.1, Avaya Aura®
Communication Manager Evolution Server R6.0.1, Avaya Aura® Session Border Controller
R6.0 and various Avaya endpoints.
Level 3 is a member of the Avaya DevConnect Service Provider program. Information in
these Application Notes has been obtained through DevConnect compliance testing and
additional technical discussions. Testing was conducted via the DevConnect Program at the
Avaya Solution and Interoperability Test Lab.
ACM; Reviewed:
SPOC 11/3/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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Table of Contents
1.
Introduction .................................................................................................................4
2.
Test Scope and Results ...............................................................................................4
2.1.
Interoperability Compliance Testing ..........................................................................4
2.2.
Test Results .................................................................................................................5
2.3.
Support ........................................................................................................................6
3.
Reference Configuration .............................................................................................7
4.
Equipment and Software Validated ............................................................................9
5.
Configure Avaya Aura® Communication Manager ...................................................9
5.1.
Licensing and Capacity .............................................................................................10
5.2.
System Features ........................................................................................................11
5.3.
IP Node Names .........................................................................................................12
5.4.
Codecs .......................................................................................................................12
5.5.
IP Network Region ...................................................................................................13
5.6.
Signaling Group ........................................................................................................14
5.7.
Trunk Group..............................................................................................................16
5.8.
Calling Party Information .........................................................................................19
5.9.
Outbound Routing.....................................................................................................20
6.
Configure Avaya Aura® Session Manager ..............................................................23
6.1.
System Manager Login and Navigation ...................................................................24
6.2.
Specify SIP Domain ..................................................................................................26
6.3.
Add Location ............................................................................................................26
6.4.
Add Adaptation Module ...........................................................................................29
6.5.
Add SIP Entities........................................................................................................32
6.6.
Add Entity Links .......................................................................................................36
6.7.
Add Routing Policies ................................................................................................38
6.8.
Add Dial Patterns ......................................................................................................40
6.9.
Add/View Session Manager .....................................................................................43
7.
Configure Avaya Aura® Session Border Controller ................................................44
7.1.
Installation Wizard ....................................................................................................45
7.1.1. Network Settings .............................................................................................................45
7.1.2. VPN Access ....................................................................................................................46
7.1.3. SBC .................................................................................................................................47
7.1.4. Confirm Installation ........................................................................................................49
7.2.
Post Installation Configuration .................................................................................50
7.2.1. Options Frequency ..........................................................................................................51
7.2.2. Blocked Headers .............................................................................................................52
7.2.3. Max-Forwards Value ......................................................................................................54
7.2.4. Diversion Header Domain Mapping ...............................................................................56
7.2.5. Remote-Party-ID Header ................................................................................................57
7.2.6. Request-URI and other Header Modifications ...............................................................59
7.2.7. Change Transport from UDP to TCP for Service Provider ............................................67
7.2.8. Third Party Call Control .................................................................................................70
7.2.9. Save the Configuration ...................................................................................................71
8.
Level 3 SIP Trunking Configuration ........................................................................71
9.
Verification and Troubleshooting .............................................................................72
ACM; Reviewed:
SPOC 11/3/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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10.
11.
12.
Conclusion ................................................................................................................73
References .................................................................................................................73
Appendix A: Avaya Aura® Session Border Controller Configuration File .........74
ACM; Reviewed:
SPOC 11/3/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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1. Introduction
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between Level 3 SIP Trunking and an Avaya SIP-enabled enterprise solution. The
Avaya solution consists of Avaya Aura® Session Manager R6.1, Avaya Aura® Communication
Manager Evolution Server R6.0.1, Avaya Aura® Session Border Controller (AA-SBC) R6.0 and
various Avaya endpoints.
Customers using this Avaya SIP-enabled enterprise solution with Level 3 SIP Trunking are able
to place and receive PSTN calls via a broadband WAN connection and the SIP protocol. This
converged network solution is an alternative to traditional PSTN trunks such as analog and/or
ISDN-PRI.
2. Test Scope and Results
A simulated enterprise site using Communication Manager, Session Manager and the AA-SBC
was connected to the public Internet using a broadband connection. The enterprise site was
configured to connect to Level 3 SIP Trunking.
2.1. Interoperability Compliance Testing
To verify SIP trunking interoperability, the following features and functionality were covered
during the interoperability compliance test:
Response to SIP OPTIONS queries
Incoming PSTN calls to various phone types
Phone types included H.323, SIP, digital, and analog telephones at the enterprise. All
inbound PSTN calls were routed to the enterprise across the SIP trunk from the service
provider.
Outgoing PSTN calls from various phone types
Phone types included H.323, SIP, digital, and analog telephones at the enterprise. All
outbound PSTN calls were routed from the enterprise across the SIP trunk to the service
provider.
Inbound and outbound PSTN calls to/from Avaya one-X® Communicator (soft client)
Avaya one-X® Communicator supports two modes (Road Warrior and Telecommuter).
Each supported mode was tested. Avaya one-X® Communicator also supports two
Voice Over IP (VoIP) protocols: H.323 and SIP. Both protocols were tested.
Various call types including: local, long distance, international, outbound toll-free,
operator assisted calls, and local directory assistance (411).
Codec G.711MU and G.729A.
DTMF transmission using RFC 2833
Caller ID presentation and Caller ID restriction
Response to incomplete call attempts and trunk errors
Voicemail navigation for inbound and outbound calls
User features such as hold and resume, internal call forwarding, transfer, and conference
Off-net call forwarding, transfer, conference and mobility (extension to cellular)
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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ACM; Reviewed:
SPOC 11/3/2011
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Items not supported or not tested included the following:
Inbound toll-free and emergency calls are supported but were not tested.
Call redirection requested by a 302 response is not supported by Level 3.
Network Call Redirection using SIP REFER containing UUI (User-To-User Information)
is not supported by Level 3.
2.2. Test Results
Interoperability testing of Level 3 SIP Trunking was completed with successful results for all test
cases with the exception of the observations/limitations noted below.
Max-Forwards:
On incoming PSTN calls to an enterprise SIP phone, the Max-Forwards
value in the incoming SIP INVITE was too small to allow the message to traverse all the
SIP hops internal to the enterprise to reach the SIP phone. Thus, the AA-SBC was used
to increase this value when the INVITE arrived at the AA-SBC from the network. (See
Section 7.2.3)
No Error Indication if No Matching Codec Offered:
If the Communication Manager
SIP trunk is improperly configured to have no matching codec with the service provider
and an outbound call is placed, the service provider returns a “480 Temporarily Moved”
response instead of a “488 Not Acceptable Here” response. As a result, the user
continues to hear ringing instead of fast busy or other error indication.
No Answer treatment:
For outbound calls to PSTN that are not answered, the service
provider returns a "487 Request terminated" after 3 minutes ringing and the enterprise
calling phone hears fast busy. No announcement is played to the caller in this situation.
Calling Party Number (PSTN transfers):
The calling party number displayed on the
PSTN phone is not updated to reflect the true connected party on calls that are transferred
to the PSTN. After the call transfer is complete, the PSTN phone displays the number of
the transferring party and not the actual connected party. Communication Manager
provides the new connected party information in the Refer-To header in the REFER
message. Level 3 does not use the Refer-To header for this purpose.
Outbound Calling Party Number (CPN) Block:
To support outbound privacy calls
(calling party number blocking), Communication Manager sends “Anonymous” as the
calling number in the SIP From header and uses the P-Asserted-Identity (PAI) header to
pass the actual calling party number. Level 3 does not support use of the PAI header for
this purpose so these calls were rejected (Level 3 returns a "604 Does not exist anywhere"
message to the enterprise; the enterprise caller hears a beep then fast-busy tones). The
workaround as documented in
Section 7.2.5
is to configure AA-SBC to add a Remote-
Party-ID header in the outbound call INVITE replicating contents from the PAI header.
In addition this functionality is available directly from Level 3 using network feature
access codes to enable or disable CPN blocking on a call-by-call basis but was not tested.
SIP 302 Redirect Method is not supported:
When a Communication Manager vector is
programmed to redirect an inbound call to a PSTN number before answering the call in
the vector with Network Call Redirection activated, Level 3 SIP Trunk will send an ACK
to the “302 Moved Temporarily” SIP message from the enterprise but will not redirect
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
5 of 82
Lev3TrkSM-AASBC
ACM; Reviewed:
SPOC 11/3/2011
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