Application Notes for Configuring Windstream SIP Trunking

Application Notes for Configuring Windstream SIP Trunking
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Windstream SIP
Trunking (Metaswitch Platform) with Avaya Aura®
Communication Manager Access Element 5.2.1, Avaya
Aura® Session Manager 6.2, and Avaya Session Border
Controller for Enterprise 4.0.5 – Issue 1.0
Abstract
These Application Notes describes the steps to configure Session Initiation Protocol (SIP)
Trunking between Windstream and Avaya Aura® Communication Manager Access Element
5.2.1, Avaya Aura® Session Manager 6.2, and Avaya Session Border Controller for Enterprise
4.0.5.
Windstream SIP Trunking provides PSTN access via a SIP trunk between the enterprise and
the Windstream network as an alternative to legacy analog or digital trunks. This approach
generally results in lower cost for the enterprise.
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.
NOTE: This Application Note is applicable with Avaya Aura® 6.2 which is currently in
Controlled Introduction. Avaya Aura® 6.2 will be Generally Available in Summer 2012.
DDT; Reviewed:
SPOC 6/5/2012
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©2012 Avaya Inc. All Rights Reserved.
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Table of Contents
1.
2.
2.1.
2.2.
2.3.
3.
4.
5.
5.1.
5.2.
5.3.
5.4.
5.5.
5.6.
5.7.
5.8.
5.9.
5.10.
5.11.
5.12.
6.
6.1.
6.2.
6.3.
6.4.
6.5.
6.6.
6.7.
6.8.
6.9.
7.
7.1.
7.1.1.
7.1.2.
7.1.3.
7.1.4.
7.1.5.
7.2.
7.2.1.
7.2.2.
7.2.3.
DDT; Reviewed:
SPOC 6/5/2012
Introduction .................................................................................................................. 4
General Test Approach and Test Results ................................................................... 4
Interoperability Compliance Testing .......................................................................... 4
Test Results .................................................................................................................. 5
Support ......................................................................................................................... 6
Reference Configuration ............................................................................................. 6
Equipment and Software Validated ............................................................................ 8
Configure Avaya Aura® Communication Manager ................................................. 9
Licensing and Capacity ............................................................................................... 9
System Features ......................................................................................................... 10
IP Node Names .......................................................................................................... 11
Codecs ........................................................................................................................ 11
IP Interface for procr ................................................................................................. 12
IP Network Region .................................................................................................... 13
Signaling Group ......................................................................................................... 14
Trunk Group ............................................................................................................... 16
Inbound Routing ........................................................................................................ 18
Calling Party Information.......................................................................................... 19
Outbound Routing ..................................................................................................... 20
Saving Communication Manager Configuration Changes ..................................... 23
Configure Avaya Aura® Session Manager.............................................................. 24
Avaya Aura® System Manager Login and Navigation .......................................... 24
Specify SIP Domain .................................................................................................. 25
Add Location.............................................................................................................. 26
Adaptations ................................................................................................................ 30
Add SIP Entities......................................................................................................... 32
Add Entity Links ........................................................................................................ 36
Add Routing Policies ................................................................................................. 37
Add Dial Patterns ....................................................................................................... 38
Verify Avaya Aura® Session Manager Instance ..................................................... 42
Configure Avaya Session Border Controller for Enterprise ................................... 44
Global Profiles ........................................................................................................... 47
Routing Profile .............................................................................................................. 47
Topology Hiding Profile .............................................................................................. 48
Server Interworking Profile.......................................................................................... 52
Signaling Manipulation ................................................................................................ 60
Server Configuration .................................................................................................... 62
Domain Policies ......................................................................................................... 70
Media Rules .................................................................................................................. 70
Signaling Rules ............................................................................................................. 72
Application Rules ......................................................................................................... 74
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7.2.4.
7.3.
7.3.1.
7.3.2.
7.3.3.
7.3.4.
8.
9.
9.1.
9.2.
10.
11.
DDT; Reviewed:
SPOC 6/5/2012
Endpoint Policy Group ................................................................................................. 75
Device Specific Settings............................................................................................ 77
Network Management .................................................................................................. 77
Signaling Interface ........................................................................................................ 78
Media Interface ............................................................................................................. 79
End Point Flows - Server Flow .................................................................................... 80
Windstream SIP Trunking Configuration ................................................................ 82
Verification and Troubleshooting ............................................................................. 83
Verification ................................................................................................................ 83
Troubleshooting ......................................................................................................... 84
Conclusion.................................................................................................................. 87
References .................................................................................................................. 87
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1. Introduction
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between Windstream and Avaya Aura® Communication Manager Access Element
5.2.1, Avaya Aura® Session Manager 6.2, and Avaya Session Border Controller for Enterprise
4.0.5.
The Windstream SIP Trunking service referenced within these Application Notes is positioned
for customers that have an IP-PBX or IP-based network equipment with SIP functionality, but
need a form of IP transport and local services to complete their solution.
Windstream SIP Trunking will enable delivery of origination and termination of local, longdistance and toll-free traffic across a single broadband connection. A SIP signaling interface will
be enabled to the Customer Premises Equipment (CPE).
2. General Test Approach and Test Results
The general test approach was to configure a simulated enterprise site using Communication
Manager, Session Manager and the Avaya Session Border Controller for Enterprise to connect to
the public Internet using a broadband connection. The enterprise site was configured to connect
to the SIP Trunking service. This configuration shown in Figure 1 was used to exercise the
features and functionality listed in Section 2.1.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent
to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a DevConnect member’s solution.
2.1. Interoperability Compliance Testing
To verify SIP trunking interoperability, the following features and functionality were covered
during the interoperability compliance test:




Incoming PSTN calls to various phone types. Phone types included H.323, 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, 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. H.323 was the only protocol tested.
DDT; Reviewed:
SPOC 6/5/2012
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©2012 Avaya Inc. All Rights Reserved.
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








Various call types including: local, long distance, international, outbound toll-free, and
local directory assistance (411).
G.711MU codec.
In-band DTMF.
G.711 Faxing.
Caller ID presentation and Caller ID restriction.
Voicemail navigation for inbound and outbound calls.
User features such as hold and resume, transfer, and conference.
Network Call Redirection using the SIP REFER method or a 302 response.
Off-net call forwarding and mobility (extension to cellular).
Items not supported or not tested included the following:
 Inbound toll-free, operator assisted calls and emergency calls (911) are supported but
were not tested as part of the compliance test.
 T.38 Fax not supported.
2.2. Test Results
Interoperability testing of Windstream SIP Trunking was completed with successful results for
all test cases with the exception of the observations/limitations described below.





T.38 Fax – The use of T.38 Fax did not pass compliance testing. Windstream returns a
“488 Not Acceptable Here” response to the SIP INVITE with T.38 parameters. Thus, the
use of T.38 Fax is not recommended with this solution.
Outbound call to busy number – When a call is placed to a PSTN number that is busy,
the caller will hear a busy tone, but Windstream will not return a “486 Busy Here”,
instead the call is answered with a “200 OK” response and a busy tone is played in the
RTP stream.
Network Call Redirection using REFER with redirected party Busy – In the testing
environment, when an inbound call was made to the enterprise, to a vector redirecting the
call to another PSTN endpoint that was busy, the caller will hear a busy tone, but
Windstream will not return a “486 Busy Here”, preventing any additional processing of
the call by Communication Manager, like the routing of the call to a local agent on the
enterprise.
Network Call Redirection using REFER with transfer – When Communication
Manager is configured With the Network Call Redirection feature enabled and an
extension receives a call from a PSTN number and attempts to transfer (either
consultative or blind) the call to another PSTN extension, the transfer is successful but
the REFER will fail. This causes the Communication Manager to stay connected to both
calls for the duration of the call rather than releasing the calls back to the PSTN.
DTMF transmission using RFC 2833 – In the testing environment, DTMF transmission
was successfully transmitted in-band using the G.711MU codec rather than in the RTP
payload as specified in RFC 2833 (Reference [19]).
Windstream SIP Trunking passed compliance testing.
DDT; Reviewed:
SPOC 6/5/2012
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2.3. Support
For technical support on Windstream SIP Trunking, contact Windstream using the Customer
Service links at www.windstream.com.
3. Reference Configuration
Figure 1 illustrates a sample Avaya SIP-enabled enterprise solution connected to Windstream
SIP Trunking. This is the configuration used for compliance testing.
The Avaya components used to create the simulated customer site included:
 Communication Manager
 Communication Manager Messaging
 Session Manager
 System Manager
 Avaya Session Border Controller for Enterprise
 Avaya G450 Media Gateway
 Avaya 9600-Series IP telephones (H.323)
 Avaya 1600-Series IP telephones (H.323)
 Avaya one-X® Communicator (H.323)
 Avaya digital and analog telephones
Located at the edge of the enterprise is the Avaya Session Border Controller for Enterprise
(Avaya SBCE). It has a public side that connects to the external network and a private side that
connects to the enterprise network. All SIP and RTP traffic entering or leaving the enterprise
flows through the Avaya SBCE. In this way, the Avaya SBCE can protect the enterprise against
any SIP-based attacks. The Avaya SBCE provides network address translation at both the IP and
SIP layers. For security reasons, any actual public IP addresses used in the configuration have
been replaced with private IP addresses. Similarly, any references to real routable PSTN
numbers have also been changed to numbers that cannot be routed by the PSTN.
DDT; Reviewed:
SPOC 6/5/2012
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Figure 1: Avaya IP Telephony Network using the SIP Trunking service
A separate trunk was created between Communication Manager and Session Manager to carry
the service provider traffic. This was done so that any trunk or codec setting required by the
service provider could be applied only to this trunk and not affect other enterprise SIP traffic. In
addition, this trunk carried both inbound and outbound traffic.
For inbound calls, the calls flow from the service provider to the Avaya SBCE then to Session
Manager. Session Manager uses the configured dial patterns (or regular expressions) and routing
policies to determine the recipient (in this case Communication Manager) and on which link to
send the call. Once the call arrives at Communication Manager, further incoming call treatment,
such as incoming digit translations and class of service restrictions may be performed.
Outbound calls to the PSTN are first processed by Communication Manager and may be subject
to outbound features such as automatic route selection, digit manipulation and class of service
restrictions. Once Communication Manager selects the proper SIP trunk, the call is routed to
Session Manager. Session Manager once again uses the configured dial patterns (or regular
expressions) to determine the route to Avaya SBCE. From Avaya SBCE, the call is sent to the
Windstream SIP Trunking service.
DDT; Reviewed:
SPOC 6/5/2012
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4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Avaya IP Telephony Solution Components
Component
Release
Avaya Aura® Communication Manager
R015x.02.1.016.4 -18942
Avaya Aura® Communication Manager
R015x.02.1.016.4
Messaging
A9021rfh
C1317rff
Avaya Aura® System Manager
6.2.0.0.15669-6.2.12.9
Avaya Aura® Session Manager
6.2.0.0.620118
Avaya Session Border Controller for
4.0.5.Q09
Enterprise
Avaya G450 Media Gateway
31.22.0
Avaya 1616 IP Telephone (H.323)
Avaya one-X® Deskphone Value Edition
1.301S
Avaya 9641 IP Telephone (H.323)
Avaya one-X® Deskphone Edition 6.2009
Avaya 9630 IP Telephone (H.323)
Avaya one-X® Deskphone Edition 3.104S
Avaya 9611 IP Telephone (H.323)
Avaya one-X® Deskphone Edition 6.2009
Avaya 9608 IP Telephone (H.323)
Avaya one-X® Deskphone Edition 6.0.3
Avaya one-X® Communicator
6.1.3.09
Avaya 2420 Digital Telephone
n/a
Avaya 6210 Analog Telephone
n/a
Metaswitch
Windstream SIP Trunking Solution Components
Component
Release
7.03.00 SU 56
Table 1: Equipment and Software Tested
The specific configuration above was used for the compatibility testing.
Note: This solution will be compatible with other Avaya Server and Media Gateway platforms
running similar versions of Communication Manager and Session Manager.
DDT; Reviewed:
SPOC 6/5/2012
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5. Configure Avaya Aura® Communication Manager
This section describes the procedure for configuring Communication Manager for the
Windstream SIP Trunking service. A SIP trunk is established between Communication Manager
and Session Manager for use by signaling traffic to and from Windstream. It is assumed the
general installation of Communication Manager, Avaya G450 Media Gateway and Session
Manager has been previously completed and is not discussed here.
The Communication Manager configuration was performed using the System Access Terminal
(SAT). Some screens in this section have been abridged and highlighted for brevity and clarity
in presentation.
Note: IP addresses and phone numbers shown throughout these Application Notes have been
edited so that the actual IP addresses of the network elements and public PSTN numbers are not
revealed.
5.1. Licensing and Capacity
Use the display system-parameters customer-options command to verify that the Maximum
Administered SIP Trunks value on Page 2 is sufficient to support the desired number of
simultaneous SIP calls across all SIP trunks at the enterprise including any trunks to the service
provider. The example shows that 450 SIP trunk licenses are available and 265 are in use. The
license file installed on the system controls the maximum values for these attributes. If a
required feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales
representative to add additional capacity.
display system-parameters customer-options
OPTIONAL FEATURES
Page
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
Maximum Administered Ad-hoc Video Conferencing Ports:
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum Media Gateway VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
DDT; Reviewed:
SPOC 6/5/2012
450
450
0
0
68
450
450
450
450
450
80
0
50
0
0
300
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USED
18
3
0
0
0
0
0
0
265
0
0
0
1
0
0
0
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5.2. System Features
Use the change system-parameters features command to set the Trunk-to-Trunk Transfer
field to all to allow incoming calls from the PSTN to be transferred to another PSTN endpoint.
If for security reasons, incoming calls should not be allowed to transfer back to the PSTN then
leave the field set to none.
change system-parameters features
Page
FEATURE-RELATED SYSTEM PARAMETERS
Self Station Display Enabled? n
Trunk-to-Trunk Transfer: all
Automatic Callback with Called Party Queuing? n
Automatic Callback - No Answer Timeout Interval (rings): 3
Call Park Timeout Interval (minutes): 10
Off-Premises Tone Detect Timeout Interval (seconds): 20
AAR/ARS Dial Tone Required? y
1 of
18
On Page 9 verify that a text string has been defined to replace the Calling Party Number (CPN)
for restricted or unavailable calls. This text string is entered in the two fields highlighted below.
The compliance test used the value of anonymous for both types of calls.
change system-parameters features
FEATURE-RELATED SYSTEM PARAMETERS
Page
9 of
18
CPN/ANI/ICLID PARAMETERS
CPN/ANI/ICLID Replacement for Restricted Calls: anonymous
CPN/ANI/ICLID Replacement for Unavailable Calls: anonymous
DISPLAY TEXT
Identity When Bridging: principal
User Guidance Display? n
Extension only label for Team button on 96xx H.323 terminals? n
INTERNATIONAL CALL ROUTING PARAMETERS
Local Country Code: 1
International Access Code: 011
ENBLOC DIALING PARAMETERS
Enable Enbloc Dialing without ARS FAC? n
CALLER ID ON CALL WAITING PARAMETERS
Caller ID on Call Waiting Delay Timer (msec): 200
DDT; Reviewed:
SPOC 6/5/2012
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5.3. IP Node Names
Use the change node-names ip command to verify that node names have been previously
defined for the IP addresses of Communication Manager (procr) and for Session Manager
(SM62). These node names will be needed for defining the service provider signaling group in
Section 5.7.
change node-names ip
Page
1 of
2
IP NODE NAMES
Name
CMM
SM62
default
procr
IP Address
10.64.19.56
10.64.90.109
0.0.0.0
10.64.19.55
5.4. Codecs
Use the change ip-codec-set command to define a list of codecs to use for calls between the
enterprise and the service provider. For the compliance test, ip-codec-set 2 was used for this
purpose. In the example below, G.711MU was entered in the Audio Codec column of the table.
Default values can be used for all other fields.
change ip-codec-set 2
Page
1 of
2
Page
2 of
2
IP Codec Set
Codec Set: 2
Audio
Codec
1: G.711MU
2:
3:
4:
Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20
Since T.38 fax is not supported, set the Fax Mode to off on Page 2.
change ip-codec-set 2
IP Codec Set
Allow Direct-IP Multimedia? n
FAX
Modem
TDD/TTY
Clear-channel
Clear-channel
DDT; Reviewed:
SPOC 6/5/2012
Mode
off
off
US
n
n
Redundancy
0
0
3
0
0
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5.5. IP Interface for procr
The add ip-interface procr or change ip-interface procr command can be used to configure
the Processor Ethernet (PE) parameters. The following screen shows the parameters used in the
sample configuration. While the focus here is the use of the PE for SIP Trunk Signaling, observe
that the Processor Ethernet will also be used for registrations from H.323 IP Telephones and
H.248 gateways in the sample configuration.
change ip-interface procr
Page
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1
IP INTERFACES
Type: PROCR
Target socket load: 1700
Enable Interface? y
Allow H.323 Endpoints? y
Allow H.248 Gateways? y
Gatekeeper Priority: 5
Network Region: 1
IPV4 PARAMETERS
Node Name: procr
Subnet Mask: /24
DDT; Reviewed:
SPOC 6/5/2012
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5.6. IP Network Region
Create a separate IP network region for the service provider trunk. This allows for separate
codec or quality of service settings to be used (if necessary) for calls between the enterprise and
the service provider versus calls within the enterprise or elsewhere. For the compliance test, IPnetwork-region 2 was chosen for the service provider trunk. IP network region 1 is the default IP
network region and encompasses the rest of the enterprise. Use the change ip-network-region 2
command to configure region 2 with the following parameters:






Set the Location field to match the enterprise location for this SIP trunk.
Set the Authoritative Domain field to match the SIP domain of the enterprise. In this
configuration, the domain name is avayalab.com. This name appears in the “From”
header of SIP messages originating from this IP region.
Enter a descriptive name in the Name field.
Enable IP-IP Direct Audio (shuffling) to allow audio traffic to be sent directly between
IP endpoints without using media resources in the Avaya Media Gateway. To enable
shuffling, set both Intra-region and Inter-region IP-IP Direct Audio fields to yes.
This is the default setting. Shuffling can be further restricted at the trunk level on the
Signaling Group form.
Set the Codec Set field to the IP codec set defined in Section 5.4.
Default values can be used for all other fields.
change ip-network-region 2
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IP NETWORK REGION
Region: 2
Location: 1
Authoritative Domain: avayalab.com
Name: SIP TRUNK
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 2
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048
IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 46
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5
AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS
RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
DDT; Reviewed:
SPOC 6/5/2012
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On Page 4, define the IP codec set to be used for traffic between region 2 and region 1 (the rest
of the enterprise). Enter the desired IP codec set in the codec set column of the row with
destination region (dst rgn) 1. Default values may be used for all other fields. The example
below shows the settings used for the compliance test. It indicates that codec set 2 will be used
for calls between region 2 (the service provider region) and region 1 (the rest of the enterprise).
change ip-network-region 2
Source Region: 2
Page
Inter Network Region Connection Management
4 of
I
G
A
R
n
A
G
L
20
M
t
c
e
t
dst codec direct
WAN-BW-limits
Video
Intervening
Dyn
rgn set
WAN Units
Total Norm Prio Shr Regions
CAC
1
2
y
NoLimit
2
2
3
4
5
6
5.7. 7Signaling Group
8
Use the
9 add signaling-group command to create a signaling group between Communication
10 and
2 Session Manager for use by the service provider trunk. This signaling group
all
Manager
is
11
used for inbound and outbound calls between the service provider and the enterprise. For the
compliance test, signaling group 1 was used for this purpose and was configured using the
parameters highlighted below.









Set the Group Type field to sip.
Set the IMS Enabled field to n. This specifies Communication Manager will serve as
an Access Element Server for Session Manager.
Set the Transport Method to the recommended default value of tls (Transport Layer
Security). Set the Near-end Listen Port and Far-end Listen Port to a valid unused
port. For ease of troubleshooting, the compliance test was conducted with the
Transport Method set to tcp and the Near-end Listen Port and Far-end Listen Port
set to 5060.
Set the Peer Detection Enabled field to y. The Peer Server field will initially be set to
Others and cannot be changed via administration. The Peer Server field will
automatically change to SM once Communication Manager detected a Session Manager
peer.
Set the Near-end Node Name to procr. This node name maps to the IP address of
Communication Manager as defined in Section 5.3.
Set the Far-end Node Name to SM. This node name maps to the IP address of Session
Manager as defined in Section 5.3.
Set the Far-end Network Region to the IP network region defined for the service
provider in Section 5.6.
Set the Far-end Domain to the domain of the enterprise.
Set Direct IP-IP Audio Connections to y. This field will enable media shuffling on the
SIP trunk.
DDT; Reviewed:
SPOC 6/5/2012
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

Set the DTMF over IP field to in-band. This value sends the DTMF digits in the RTP
audio stream.
Default values may be used for all other fields.
add signaling-group 1
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SIGNALING GROUP
Group Number: 1
Group Type: sip
Transport Method: tcp
IMS Enabled? n
IP Video? n
Near-end Node Name: procr
Near-end Listen Port: 5060
Far-end Node Name: SM62
Far-end Listen Port: 5060
Far-end Network Region: 2
Far-end Domain: avayalab.com
Incoming Dialog Loopbacks: eliminate
DTMF over IP: in-band
Session Establishment Timer(min): 3
Enable Layer 3 Test? n
H.323 Station Outgoing Direct Media? n
DDT; Reviewed:
SPOC 6/5/2012
Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
Direct IP-IP Early Media?
Alternate Route Timer(sec):
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n
n
y
n
n
6
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5.8. Trunk Group
Use the add trunk-group command to create a trunk group for the signaling group created in
Section 5.7. For the compliance test, trunk group 1 was configured using the parameters
highlighted below.








Set the Group Type field to sip.
Enter a descriptive name for the Group Name.
Enter an appropriate Class of Restriction (COR) designated for SIP Trunks in the COR
field.
Enter an available trunk access code (TAC) that is consistent with the existing dial plan
in the TAC field.
Set the Service Type field to public-ntwrk.
Set the Signaling Group to the signaling group shown in the previous step.
Set the Number of Members field to the number of trunk members in the SIP trunk
group. This value determines how many simultaneous SIP calls can be supported by this
trunk.
Default values were used for all other fields.
add trunk-group 1
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21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
1
Group Type:
SIP Trunk to SP
COR:
two-way
Outgoing Display?
n
0
public-ntwrk
Auth Code?
sip
CDR Reports: y
10
TN: 1
TAC: *101
n
Night Service:
n
Signaling Group: 1
Number of Members: 10
On Page 2, verify that the Preferred Minimum Session Refresh Interval is set to a value
acceptable to the service provider. This value defines the interval that re-INVITEs must be sent
to keep the active session alive. For the compliance test, the value of 600 seconds was used.
add trunk-group 1
Group Type: sip
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21
TRUNK PARAMETERS
Unicode Name: auto
Redirect On OPTIM Failure: 5000
SCCAN? n
Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 600
Disconnect Supervision - In? y
Out? y
auto
Delay
Setup When
Accessed Via IGAR? n16 of 88
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Notes
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WSSipCM521SM62
On Page 3, set the Numbering Format field to public. This field specifies the format of the
calling party number (CPN) sent to the far-end.
Set the Replace Restricted Numbers and Replace Unavailable Numbers fields to y. This will
allow the CPN displayed on local endpoints to be replaced with the value set in Section 5.2, if
the inbound call enabled CPN block. For outbound calls, these same settings request that CPN
block be activated on the far-end destination if a local user requests CPN block on a particular
call routed out this trunk.
add trunk-group 1
TRUNK FEATURES
ACA Assignment? n
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21
Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? y
Replace Unavailable Numbers? y
Show ANSWERED BY on Display? y
Modify Tandem Calling Number: no
Show ANSWERED BY on Display? y
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On Page 4, set the Network Call Redirection field to y. This allows inbound calls transferred
back to the PSTN to use the SIP REFER method, see Reference [18]. Set the Send Diversion
Header field to y. This field provides additional information to the network if the call has been
re-directed. This is necessary to support call forwarding of inbound calls back to the PSTN and
some Extension to Cellular (EC500) call scenarios. Set the Support Request History field to n.
Default values may be used for all other fields
add trunk-group 1
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21
PROTOCOL VARIATIONS
Mark Users as Phone?
Prepend '+' to Calling Number?
Send Transferring Party Information?
Network Call Redirection?
Send Diversion Header?
Support Request History?
Telephone Event Payload Type:
n
n
n
y
y
n
5.9. Inbound Routing
In general, the incoming call handling treatment for a trunk group can be used to manipulate the
digits received for an incoming call if necessary. Since Session Manager is present, Session
Manager can be used to perform digit conversion using an Adaptation, and digit manipulation
via the Communication Manager incoming call handling table may not be necessary. If the DID
number sent by Windstream is unchanged by Session Manager, then the DID number can be
mapped to an extension using the incoming call handling treatment of the receiving trunk group.
Use the change inc-call-handling-trmt trunk-group command to create an entry for each DID.
As an example, the following screen illustrates a conversion of DID number 5015551490 to
extension 19000. Both Session Manager digit conversion and Communication Manager
incoming call handling treatment methods were tested successfully.
change inc-call-handling-trmt trunk-group 1
INCOMING CALL HANDLING TREATMENT
Service/
Number
Number
Del Insert
Feature
Len
Digits
public-ntwrk
10 5015551490
10 19000
public-ntwrk
DDT; Reviewed:
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5.10. Calling Party Information
The calling party number is sent in the SIP “From”, “Contact” and “PAI” headers. Since public
numbering was selected to define the format of this number (Section 5.8), use the change
public-unknown-numbering command to create an entry for each extension which has a DID
assigned. The DID number will be one assigned by the SIP service provider. It is used to
authenticate the caller.
In the bolded rows shown in the example abridged output below, Communication Mana ger
extensions are mapped to DID numbers that are known to Windstream for this SIP Trunk
connection when the call uses trunk group 1.
change public-unknown-numbering 1
Page
1 of
NUMBERING - PUBLIC/UNKNOWN FORMAT
Total
Ext Ext
Trk
CPN
CPN
Len Code
Grp(s)
Prefix
Len
Total Administered: 6
5 12200
1
5015551070
10
Maximum Entries: 240
5 12201
1
5015551071
10
5 12202
1
5015551072
10
5 12203
1
5015551073
10
5 12204
1
5015551074
10
5 12205
1
5015551075
10
DDT; Reviewed:
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5.11. Outbound Routing
In these Application Notes, the Automatic Route Selection (ARS) feature is used to route
outbound calls via the SIP trunk to the service provider. In the sample configuration, the single
digit 9 is used as the ARS access code. Enterprise callers will dial 9 to reach an outside line.
This common configuration is illustrated below. Use the change dialplan analysis command to
define a dialed string beginning with 9 of length 1 as a feature access code (fac).
change dialplan analysis
Page
DIAL PLAN ANALYSIS TABLE
Location: all
Dialed
String
1
2
4
5
6
7
8
9
*
#
Total
Length
5
5
4
4
5
4
1
1
4
4
Call
Type
ext
ext
ext
ext
ext
ext
fac
fac
dac
fac
Dialed
String
Total Call
Length Type
1 of
Percent Full:
Dialed
String
12
0
Total Call
Length Type
Use the change feature-access-codes command to configure 9 as the Auto Route Selection
(ARS) – Access Code 1.
change feature-access-codes
Page
1 of
FEATURE ACCESS CODE (FAC)
Abbreviated Dialing List1 Access Code: #110
Abbreviated Dialing List2 Access Code: #111
Abbreviated Dialing List3 Access Code: #112
Abbreviated Dial - Prgm Group List Access Code: #113
Announcement Access Code: #114
Answer Back Access Code:
Attendant Access Code:
Auto Alternate Routing (AAR) Access Code: 8
Auto Route Selection (ARS) - Access Code 1: 9
Access Code 2:
Automatic Callback Activation:
Deactivation:
Call Forwarding Activation Busy/DA: #002
All:
Deactivation: #004
Call Forwarding Enhanced Status:
Act:
Deactivation:
Call Park Access Code: *40
Call Pickup Access Code: *41
CAS Remote Hold/Answer Hold-Unhold Access Code: *42
CDR Account Code Access Code:
Change COR Access Code:
Change Coverage Access Code:
Conditional Call Extend Activation:
Deactivation:
Contact Closure
Open Code: *80
Close Code: #80
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Use the change ars analysis command to configure the routing of dialed digits following the
first digit 9.





Dialed String: enter the leading digits (e.g., 1303) necessary to uniquely select the
desired route pattern.
Total Min: enter the minimum number of digits (e.g., 11) expected for this PSTN
number.
Total Max: enter the maximum number of digits (e.g., 11) expected for this PSTN
number.
Route Pattern: enter the route pattern number (e.g., 1) to be used. The route pattern (to
be defined next) will specify the trunk group(s) to be used for calls matching the dialed
number.
Call Type: enter fnpa, the call type for North American 1+10 digit calls. For local 7 or
10 digit calls enter hnpa. For 411 and 911 calls use svcl and emer respectively. The call
type tells Communication Manager what kind of call is made to help decide how to
handle the dialed string and whether or not to include a preceding 1. For more
information and a complete list of Communication Manager call types, see Reference [3]
and [4].
The example below shows a subset of the dialed strings tested as part of the compliance test. See
Section 2.1Error! Reference source not found. for the complete list of call types tested. All
dialed strings are mapped to route pattern 1 which contains the SIP trunk to the service provider
(as defined next).
change ars analysis 1
Page
ARS DIGIT ANALYSIS TABLE
Location: all
Dialed
String
12
13
14
15
16
17
18
19
303
411
501
720
911
DDT; Reviewed:
SPOC 6/5/2012
Total
Min Max
11
11
11
11
11
11
11
11
11
11
11
11
11
11
11
11
10
10
3
3
10
10
10
10
3
3
Route
Pattern
1
1
1
1
1
1
1
1
1
1
1
1
1
Call
Type
fnpa
fnpa
fnpa
fnpa
fnpa
fnpa
fnpa
fnpa
hnpa
svcl
hnpa
hnpa
emer
Node
Num
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
1 of
Percent Full:
2
0
ANI
Reqd
n
n
n
n
n
n
n
n
n
n
n
n
n
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The route pattern defines which trunk group will be used for the call and performs any necessary
digit manipulation. Use the change route-pattern command to configure the parameters for the
service provider trunk route pattern in the following manner. The example below shows the
values used for route pattern 1 during the compliance test.




Pattern Name: Enter a descriptive name.
Grp No: Enter the outbound trunk group for the SIP service provider. For the
compliance test, trunk group 1 was used.
FRL: Set the Facility Restriction Level (FRL) field to a level that allows access to
this trunk for all users that require it. The value of 0 is the least restrictive level.
Pfx Mrk: 1 The prefix mark (Pfx Mrk) of 1 will prefix any FNPA 10-digit number
with a 1 and leave numbers of any other length unchanged. This will ensure 1 + 10
digits are sent to the service provider for long distance North American Numbering
Plan (NANP) numbers. All HNPA 10 digit numbers are left unchanged.
change route-pattern 1
Page
1 of
Pattern Number: 1
Pattern Name: WINDSTREAM SIP TRK
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
DCS/
No
Mrk Lmt List Del Digits
QSIG
Dgts
Intw
1: 1
0
1
n
2:
n
3:
n
4:
n
5:
n
6:
n
1:
2:
3:
4:
5:
6:
BCC VALUE TSC CA-TSC
0 1 2 M 4 W
Request
ITC BCIE Service/Feature PARM
y
y
y
y
y
y
rest
rest
rest
rest
rest
rest
y
y
y
y
y
y
y
y
y
y
y
y
DDT; Reviewed:
SPOC 6/5/2012
y
y
y
y
y
y
y
y
y
y
y
y
n
n
n
n
n
n
n
n
n
n
n
n
3
IXC
user
user
user
user
user
user
No. Numbering LAR
Dgts Format
Subaddress
none
none
none
none
none
none
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Use the change ars digit-conversion command to manipulate the routing of dialed digits that
match the DIDs to prevent these calls from going out the PSTN and using unnecessary SIP trunk
resources. The example below shows the DID numbers assigned by Windstream being
converted to 5 digit extensions.
change ars digit-conversion 0
ARS DIGIT CONVERSION TABLE
Location: all
Page
1 of
Percent Full: 0
Matching Pattern
Min
Max
Del
Replacement String
Net
5015511071
5015551070
5015551072
5015551073
5015551074
5015551075
10
10
10
10
10
10
10
10
10
10
10
10
10
10
10
10
10
10
12201
12200
12202
12203
12204
12205
ext
ext
ext
ext
ext
ext
Conv ANI Req
y
y
y
y
y
y
5.12. Saving Communication Manager Configuration Changes
y y y y y n
n
rest
n
n
all
n
n
n
n
n
n
n
n
n
n
n
n
n
n
n
n
n
n 4:
none
none
none
none
none
none
The5:command
all rest
can be used to save the configuration.
y y y y save
y n translation
n
6: y y y y y n
4: y y y y y n
save translation
5: y y y y y n
6: y y y y y n
2
rest
rest
rest
rest
SAVE TRANSLATION
Command Completion Status
Error Code
Success
0
DDT; Reviewed:
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6. Configure Avaya Aura® Session Manager
This section provides the procedures for configuring Session Manager. The procedures include
adding the following items:




SIP domain
Logical/physical Location that can be occupied by SIP Entities
SIP Entities corresponding to Communication Manager, Avaya SBCE and Session Manager
Entity Links, which define the SIP trunk parameters used by Session Manager when routing
calls to/from SIP Entities
 Routing Policies, which control call routing between the SIP Entities
 Dial Patterns, which govern to which SIP Entity a call is routed
 Session Manager Instance, corresponding to the Session Manager server to be administered
in System Manager.
It may not be necessary to create all the items above when creating a connection to the service
provider since some of these items would have already been defined as part of the initial Session
Manager installation. This includes items such as certain SIP domains, locations, SIP entities,
and Session Manager itself. However, each item should be reviewed to verify the configuration.
6.1. Avaya Aura® System Manager Login and Navigation
Session Manager configuration is accomplished by accessing the browser-based GUI of System
Manager, using the URL https://<ip-address>/SMGR, where <ip-address> is the IP address of
System Manager. Log in with the appropriate credentials and click on Log On (not shown).
The screen shown below is then displayed.
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Most of the configuration items are performed in the Routing Element. Click on Routing in the
Elements column shown above to bring up the Introduction to Network Routing Policy screen.
6.2. Specify SIP Domain
Create a SIP domain for each domain for which Session Manager will need to be aware in order
to route calls. For the compliance test, this includes the enterprise domain (avayalab.com).
Navigate to Routing  Domains and click the New button in the right pane (not shown). In the
new right pane that appears, fill in the following:
 Name:
 Type:
 Notes:
Enter the domain name.
Select sip from the pull-down menu.
Add a brief description (optional).
Click Commit. The screen below shows the entry for the avayalab.com domain.
DDT; Reviewed:
SPOC 6/5/2012
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6.3. Add Location
Locations can be used to identify logical and/or physical locations where SIP Entities reside for
purposes of bandwidth management and call admission control. To add a location, navigate to
Routing Locations in the left-hand navigation pane and click the New button in the right pane
(not shown).
In the General section, enter the following values. Use default values for all remaining fields:
 Name: Enter a descriptive name for the location.
 Notes: Add a brief description (optional).
The Location Pattern was not populated. The Location Pattern is used to identify call routing
based on IP address. Session Manager matches the IP address against the patterns defined in this
section. If a call is from a SIP Entity that does not match the IP address pattern then Session
Manager uses the location administered for the SIP Entity. In this sample configuration
Locations are added to SIP Entities (Section 6.5), so it was not necessary to add a pattern.
The following screen shows the addition of Location_1, this location will be used for Session
Manager. Click Commit to save.
DDT; Reviewed:
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Note: Call bandwidth management parameters should be set per customer requirement.
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Repeat the preceding procedure to create a separate Location for Communication Manager and
Avaya SBCE. Displayed below is the screen for CM521 used for Communication Manager.
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Below is the screen for AvayaSBCE used for Avaya SBCE.
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6.4. Adaptations
To view or change adaptations, select Routing  Adaptations. Click on the checkbox
corresponding to the name of an adaptation and Edit to edit an existing adaptation, or the New
button to add an adaptation. Click the Commit button after changes are completed.
The following screen shows the adaptation that was available in the sample configuration.
The adapter named CM521 Adaptation will later be assigned to the SIP Entity linking Session
Manager to Communication Manager for calls involving Windstream SIP Trunking. This
adaptation uses the DigitConversionAdapter to convert digits between Communication
Manager and Windstream.
DDT; Reviewed:
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Scrolling down, the following screen shows a portion of the CM521 Adaptation adapter that
can be used to convert digits between the Communication Manager extension numbers (user
extensions, VDNs) and the DID numbers assigned by Windstream.
An example portion of the settings for Digit Conversion for Outgoing Calls from SM (i.e.,
inbound to Communication Manager) is shown below.
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6.5.
Add SIP Entities
A SIP Entity must be added for Session Manager and for each SIP telephony system connected
to it which includes Communication Manager and Avaya SBCE. Navigate to Routing  SIP
Entities in the left-hand navigation pane and click on the New button in the right pane (not
shown).
In the General section, enter the following values. Use default values for all remaining fields:
 Name:
Enter a descriptive name.
 FQDN or IP Address: Enter the FQDN or IP address of the SIP Entity that is used for SIP
signaling.
 Type:
Enter Session Manager for Session Manager, CM for
Communication Manager and SIP Trunk for Avaya SBCE.
 Adaptation:
This field is only present if Type is not set to Session Manager.
If applicable, select the Adaptation Name that will be applied to
this entity.
 Location:
Select one of the locations defined previously.
 Time Zone:
Select the time zone for the location above.
The following screen shows the addition of Session Manager. The IP address of the Session
Manager signaling interface is entered for FQDN or IP Address.
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To define the ports used by Session Manager, scroll down to the Port section of the SIP Entity
Details screen. This section is only present for Session Manager SIP entities. This section
defines a default set of ports that Session Manager will use to listen for SIP requests, typically
from registered SIP endpoints. Session Manager can also listen on additional ports defined
elsewhere such as the ports specified in the SIP Entity Link definition in Section 6.6.
In the Port section, click Add and enter the following values. Use default values for all
remaining fields:
 Port:
Port number on which Session Manager can listen for SIP
requests.
 Protocol:
Transport protocol to be used to send SIP requests.
 Default Domain:
The domain used for the enterprise.
Defaults can be used for the remaining fields. Click Commit to save.
For the compliance test, two Port entries were added.
DDT; Reviewed:
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The following screen shows the addition of Communication Manager. The FQDN or IP
Address field is set to the IP address defined in Section 5.3 of the procr interface on
Communication Manager. The Adaptation field is set to the Adaptation created in Section 6.4
and the Location is set to the one defined for Communication Manager in Section 6.3.
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The following screen shows the addition of ASBCE SIP Entity. The FQDN or IP Address field
is set to the IP address of its private network interface (see Figure 1). The Location is set to the
one defined for Avaya SBCE in Section 6.3. Link Monitoring Enabled was selected for SIP
Link Monitoring using the specific time settings for Proactive Monitoring Interval (in
seconds) and Reactive Monitoring Interval (in seconds) for the compliance test. These time
settings should be adjusted or left at their default values per customer needs and requirements.
DDT; Reviewed:
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6.6. Add Entity Links
A SIP trunk between Session Manager and a telephony system is described as an Entity Link.
Two Entity Links were created; one to Communication Manager for use only by service provider
traffic and one to Avaya SBCE. To add an Entity Link, navigate to Routing  Entity Links in
the left-hand navigation pane and click on the New button in the right pane (not shown). Fill in
the following fields in the new row that is displayed:




Name:
SIP Entity 1:
Protocol:
Port:
 SIP Entity 2:
 Port:
 Trusted:
Enter a descriptive name.
Select the SIP Entity for Session Manager.
Select the transport protocol used for this link.
Port number on which Session Manager will receive SIP requests from
the far-end. For Communication Manager, this must match the
Far-end Listen Port defined on the Communication Manager signaling
group in Section 5.7.
Select the name of the other system. For Communication Manager,
select the Communication Manager SIP Entity defined in Section 6.4.
Port number on which the other system receives SIP requests from the
Session Manager. For Communication Manager, this must match the
Near-end Listen Port defined on the Communication Manager signaling
group in Section 5.7.
Check this box. Note: If this box is not checked, calls from the associated
SIP Entity specified in Section 6.5 will be denied.
Click Commit to save. The following screens illustrate the Entity Links to Communication
Manager and Avaya SBCE. It should be noted that in a customer environment the Entity Link to
Communication Manager would normally use TLS. For the compliance test, TCP was used to
aid in troubleshooting since the signaling traffic would not be encrypted.
Entity Link to Communication Manager:
DDT; Reviewed:
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Entity Link to Avaya SBCE:
6.7. Add Routing Policies
Routing policies describe the conditions under which calls will be routed to the SIP Entities
specified in Section 6.5. Two routing policies must be added; one for Communication Manager
and one for Avaya SBCE. To add a routing policy, navigate to Routing  Routing Policies in
the left-hand navigation pane and click on the New button in the right pane (not shown). The
screen below is displayed. Fill in the following:
In the General section, enter the following values. Use default values for all remaining fields:
 Name:
Enter a descriptive name.
 Notes:
Add a brief description (optional).
In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not
shown). Select the appropriate SIP entity to which this routing policy applies and click Select
(not shown). The selected SIP Entity displays on the Routing Policy Details page as shown
below. Use default values for remaining fields. Click Commit to save.
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The following screens show the Routing Policies for Communication Manager and Avaya
SBCE.
6.8. Add Dial Patterns
Dial Patterns are needed to route calls through Session Manager. For the compliance test, dial
patterns were needed to route calls from Communication Manager to Windstream and vice versa.
Dial Patterns define which route policy will be selected for a particular call based on the dialed
digits, destination domain and originating location. To add a dial pattern, navigate to Routing
 Dial Patterns in the left-hand navigation pane and click on the New button in the right pane
(not shown). Fill in the following, as shown in the screens below:
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In the General section, enter the following values. Use default values for all remaining fields:
 Pattern:
Enter a dial string that will be matched against the Request-URI of the
call.
 Min:
Enter a minimum length used in the match criteria.
 Max:
Enter a maximum length used in the match criteria.
 SIP Domain:
Enter the destination domain used in the match criteria.
 Notes:
Add a brief description (optional).
In the Originating Locations and Routing Policies section, click Add. From the Originating
Locations and Routing Policy List that appears (not shown), select the appropriate originating
location for use in the match criteria. Lastly, select the routing policy from the list that will be
used to route all calls that match the specified criteria. Click Select.
Default values can be used for the remaining fields. Click Commit to save.
Two examples of the dial patterns used for the compliance test are shown below. The first
example shows that 11 digit dialed numbers that begin with 1 originating from CM521 uses
route policy To-ASBCE.
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The second example shows that a 10 digit number starting with 501555107 to domain
avayalab.com and originating from AvayaSBCE uses route policy To-CM521. This is a DID
range 501-555-1070 through 501-555-1079 assigned to the enterprise from Windstream.
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The complete list of dial patterns defined for the compliance test is shown below.
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6.9. Add Avaya Aura® Session Manager Instance
The creation of a Session Manager Instance provides the linkage between System Manager and
Session Manager. This was most likely done as part of the initial Session Manager installation.
To add a Session Manager, navigate to Elements  Session Manager  Session Manager
Administration in the left-hand navigation pane and click on the New button in the right pane
(not shown). If the Session Manager instance already exists, click View (not shown) to view the
configuration. Enter/verify the data as described below and shown in the screen below:
In the General section, enter the following values:
 SIP Entity Name:
Select the SIP Entity created for Session
Manager.
 Description:
Add a brief description (optional).
 Management Access Point Host Name/IP: Enter the IP address of the Session Manager
management interface.
The screen below shows the Session Manager values used for the compliance test.
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In the Security Module section, enter the following values:
 SIP Entity IP Address:
Should be filled in automatically based on the SIP Entity
Name. Otherwise, enter IP address of Session Manager
signaling interface.
 Network Mask:
Enter the network mask corresponding to the IP address of
Session Manager.
 Default Gateway:
Enter the IP address of the default gateway for Session
Manager.
Use default values for the remaining fields. Click Save (not shown) to add this Session
Manager. The screen below shows the remaining Session Manager values used for the
compliance test.
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7. Configure Avaya Session Border Controller for Enterprise
This section covers the configuration of Avaya Session Border Controller for Enterprise (Avaya
SBCE). It is assumed that the software has already been installed. For additional information on
these configuration tasks, see Reference [15] and [16].
A pictorial view of this configuration is shown below. It shows the components needed for the
compliance test. Each of these components is defined in the Avaya SBCE web configuration as
described in the following sections.
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Use a WEB browser to access the UC-Sec web interface, enter https://<ip-addr>/ucsec in the
address field of the web browser, where <ip-addr> is the management LAN IP address of UCSec.
Log in with the appropriate credentials. Click Sign In.
The main page of the UC-Sec Control Center will appear.
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To view system information that was configured during installation, navigate to UC-Sec Control
Center  System Management. A list of installed devices is shown in the right pane. In the
case of the sample configuration, a single device named Sipera is shown. To view the
configuration of this device, click the monitor icon (the third icon from the right).
The System Information screen shows the Network Settings, DNS Configuration and
Management IP information provided during installation and corresponds to Figure 1. The
Box Type was set to SIP and the Deployment Mode was set to Proxy. Default values were
used for all other fields.
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7.1. Global Profiles
Global Profiles allows for configuration of parameters across all UC-Sec appliances.
7.1.1. Routing Profile
Routing profiles define a specific set of packet routing criteria that are used in conjunction with
other types of domain policies to identify a particular call flow and thereby ascertain which
security features will be applied to those packets. Parameters defined by Routing Profiles
include packet transport settings, name server addresses and resolution methods, next hop
routing information, and packet transport types.
Create a Routing Profile for Session Manager and Windstream SIP Trunk. To add a routing
profile, navigate to UC-Sec Control Center Global Profiles  Routing and select Add
Profile. Enter a Profile Name and click Next to continue (not shown).
In the new window that appears, enter the following values. Use default values for all remaining
fields:


URI Group:
Next Hop Server 1:

Next Hop Server 2:

Routing Priority Based on
Next Hop Server:
Use Next Hop for
In-Dialog Messages:


Outgoing Transport:
Select “*” from the drop down box.
Enter the Domain Name or IP address of the
Primary Next Hop server.
(Optional) Enter the Domain Name or IP address of
the secondary Next Hop server.
Checked.
Select only if there is no secondary Next Hop
server.
Choose the protocol used for transporting outgoing
signaling packets.
Click Finish (not shown).
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The following screen shows the Routing Profile to Communication Manager. The Next Hop
Server 1 IP address must match the IP address of the Session Manager Security Module in
Section 6.9. The Outgoing Transport and port number must match the Avaya SBCE Entity Link
created on Session Manager in Section 6.6.
The following screen shows the Routing Profile to Windstream. In the Next Hop Server 1 field
enter the IP address and port number that Windstream uses to listen for SIP traffic.
7.1.2. Topology Hiding Profile
The Topology Hiding profile manages how various source, destination and routing information
in SIP and SDP message headers are substituted or changed to maintain the integrity of the
network. It hides the topology of the enterprise network from external networks.
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Create a Topology Hiding Profile for the enterprise and SIP Trunk. In the sample configuration,
the Enterprise and SIP Trunk profiles were cloned from the default profile. To clone a default
profile, navigate to UC-Sec Control Center Global Profiles  Topology Hiding. Select the
default profile and click on Clone Profile as shown below.
Enter a descriptive name for the new profile and click Finish.
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Edit the Enterprise profile to overwrite the To, Request-Line and From headers shown below
to the enterprise domain. The Overwrite Value should match the Domain set in Session
Manager (Section 6.2) and the Communication Manager signaling group Far-end Domain
(Section 5.7). Click Finish to save the changes.
It is not necessary to modify the SIP Trunk profile from the default values. The following
screen shows the Topology Hiding Policy created for Windstream.
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When creating or editing Topology Hiding Profiles, there are six types of headers available for
selection in the Header drop-down list to choose from. In addition to the six headers, there are
additional headers not listed that are affected when either of two types of listed headers (e.g., To
Header and From Header) are selected in the Header drop-down list. Table 2 lists the six
headers along with all of the other affected headers in three header categories (e.g., Source
Headers, Destination Headers, and SDP Headers).
Topology Hiding Headers
Main Header Names
Header(s) Affected by Main Header
Source Headers
Record-Route
From
(1) Referred-By
(2) P-Asserted Identity
Via
Destination Headers
To
Request-Line
(1) ReferTo
SDP Headers
Origin Header
Table 2: Topology Hiding Headers
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7.1.3. Server Interworking Profile
The Server Internetworking profile configures and manages various SIP call server-specific
parameters such as TCP and UDP port assignments, heartbeat signaling parameters (for HA
deployments), DoS security statistics, and trusted domains. Interworking Profile features are
configured based on different Trunk Servers. There are default profiles available that may be
used as is, or modified, or new profiles can be configured as described below.
In the sample configuration, separate Server Interworking Profiles were created for Enterprise
and Windstream.
7.1.3.1 Server Interworking Profile – Enterprise
To create a new Server Interworking Profile for the enterprise, navigate to UC-Sec Control
Center Global Profiles  Server Interworking and click on Add Profile as shown below.
Enter a descriptive name for the new profile and click Next to continue.
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In the new window that appears, default values can be used. Click Next to continue.
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Default values can also be used for the next two windows that appear. Click Next to continue.
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On the Advanced Settings window uncheck the following default settings:
 Topology Hiding: Change Call-ID
 Change Max Forwards
Click Finish to save changes.
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7.1.3.2 Server Interworking Profile – Windstream
To create a new Server Interworking Profile for Windstream, navigate to UC-Sec Control
Center Global Profiles  Server Interworking and click on Add Profile as shown below.
Enter a descriptive name for the new profile and click Next to continue.
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In the new window that appears, default values can be used. Click Next to continue.
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Default values can also be used for the next two windows that appear. Click Next to continue.
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On the Advanced Settings window uncheck the following default settings:
 Topology Hiding: Change Call-ID
 Change Max Forwards
Click Finish to save changes.
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7.1.4. Signaling Manipulation
The Signaling Manipulation feature allows the ability to add, change and delete any of the
headers in a SIP message. This feature will add the ability to configure such manipulation in a
highly flexible manner using a proprietary scripting language called SigMa.
The SigMa scripting language is designed to express any of the SIP header manipulation
operations to be done by the Avaya SBCE. Using this language, a script can be written and tied
to a given flow through the EMS GUI. The Avaya SBCE appliance then interprets this script at
the given entry point or “hook point”.
These Application Notes will not discuss the full feature of the Signaling Manipulation but will
show an example of a script created during compliance testing to aid in topology hiding and to
remove unwanted headers in the SIP messages to Windstream. To create a new Signaling
Manipulation, navigate to UC-Sec Control Center Global Profiles  Signaling
Manipulation and click on Add Script (not shown). A new blank SigMa Editor window will
pop up. For more information on Signaling Manipulation see Reference [16].
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The following sample script begins with a comment describing what will take place in the script.
The script will act on all outbound traffic to Windstream after the SIP message has been routed
through the Avaya SBCE. The script is further broken down as follows:






within session “All”
act on message
%DIRECTION=“OUTBOUND”
Transformations applied to all SIP sessions.
Actions to be taken to any SIP message.
Applied to a message leaving the Avaya
SBCE.
%ENTRY_POINT=“POST_ROUTING” The “hook point” to apply the script after the
SIP message has routed through the Avaya
SBCE.
%HEADERS[“p-asserted-identity”][1]; Used to retrieve an entire header. The first
dimension denotes which header while the
second dimension denotes the 1 st instance of
the header in a message.
.regex_replace(“avayalab\.com”,
An action to replace a given match with the
“205.xxx.xxx.92:5060”);
provide string (e.g., find “avayalab.com”
and replace it with the external interface IP
address and port). The backslash is used to
escape the special meaning of “.” in a
Regular Expression.
With this script, the P-Location and Alert-Info headers will be removed. The P-AssertedIdentity header will be modified by replacing the domain “avayalab.com” with the external IP
address of Avaya SBCE and the SIP port of 5060.
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The following screen shows the finished Signaling Manipulation Script SIP Trunk4_Script.
7.1.5. Server Configuration
The Server Configuration screen contains four tabs: General, Authentication, Heartbeat, and
Advanced. Together, these tabs configure and manage various SIP call server-specific
parameters such as TCP and UDP port assignments, heartbeat signaling parameters, DoS security
statistics, and trusted domains.
In the sample configuration, separate Server Configurations were created for Session Manager
and Windstream.
7.1.5.1 Server Configuration – Session Manager
To add a Server Configuration Profile for Session Manager navigate to UC-Sec Control Center
Global Profiles  Server Configuration and click on Add Profile as shown below.
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Enter a descriptive name for the new profile and click Next.
In the new window that appears, enter the following values. Use default values for all remaining
fields:


Server Type:
IP Addresses /
Supported FQDNs:

Supported Transports:

TCP Port:
Select Call Server from the drop-down box.
Enter the IP address of the Session Manager signaling
interface. This should match the IP address of the Session
Manager Security Module in Section 6.9.
Select the transport protocol used to create the Avaya
SBCE Entity Link on Session Manager in Section 6.6.
Port number on which to send SIP requests to Session
Manager. This should match the port number used in the
Avaya SBCE Entity Link on Session Manager in Section
6.6.
Click Next to continue.
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Verify Enable Authentication is unchecked as Session Manager does not require authentication.
Click Next to continue.
In the new window that appears, enter the following values. Use default values for all remaining
fields:



Enabled Heartbeat:
Method:
Frequency:

From URI:

TO URI:
Checked.
Select OPTIONS from the drop-down box.
Choose the desired frequency in seconds the Avaya
SBCE will send SIP OPTIONS. For compliance
testing 60 seconds was chosen.
Enter an URI to be sent in the FROM header for
SIP OPTIONS.
Enter an URI to be sent in the TO header for SIP
OPTIONS.
Click Next to continue.
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In the new window that appears, select the Interworking Profile created for the enterprise in
Section 7.1.3.1. Use default values for all remaining fields. Click Finish to save the
configuration.
7.1.5.2 Server Configuration - Windstream
To add a Server Configuration Profile for Windstream navigate to UC-Sec Control Center
Global Profiles  Server Configuration and click on Add Profile (not shown). Enter a
descriptive name for the new profile and click Next.
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In the new window that appears, enter the following values. Use default values for all remaining
fields:


Server Type:
IP Addresses /
Supported FQDNs:

Supported Transports:

TCP Port:
Select Trunk Server from the drop-down box.
Enter the IP address(es) of the SIP proxy(ies) of the service
provider. In the case of the compliance test, this is the
Windstream SIP Trunk IP address. This will associate the
inbound SIP messages from Windstream to this Sever
Configuration.
Select the transport protocol to be used for SIP traffic
between Avaya SBCE and Windstream.
Enter the port number that Windstream uses to send SIP
traffic.
Click Next to continue.
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Verify Enable Authentication is unchecked as Windstream does not require authentication.
Click Next to continue.
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In the new window that appears, enter the following values. Use default values for all remaining
fields:



Enabled Heartbeat:
Method:
Frequency:

From URI:

TO URI:
Checked.
Select OPTIONS from the drop-down box.
Choose the desired frequency in seconds the Avaya
SBCE will send SIP OPTIONS. For compliance
testing 60 seconds was chosen.
Enter an URI to be sent in the FROM header for
SIP OPTIONS.
Enter an URI to be sent in the TO header for SIP
OPTIONS.
Click Next to continue.
The SIP OPTIONS are sent to the SIP proxy(ies) entered in the IP Addresses /Supported
FQDNs in the Server Configuration Profile. The URI of PING@windstream.com was used in
the sample configuration to better identify the SIP OPTIONS in the call traces. Windstream does
not look at the From and To headers when replying to SIP OPTIONS so any URI can be used as
long as it is in the proper format (USER@DOMAIN).
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In the new window that appears, select the Interworking Profile created for Windstream in
Section 7.1.3.2. Select the Signaling Manipulation Script created in Section 7.1.4. Use
default values for all remaining fields. Click Finish to save the configuration.
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7.2. Domain Policies
The Domain Policies feature configures, applies, and manages various rule sets (policies) to
control unified communications based upon various criteria of communication sessions
originating from or terminating in the enterprise. These criteria can be used to trigger policies
which, in turn, activate various security features of the UC-Sec security device to aggregate,
monitor, control, and normalize call flows. There are default policies available to use, or a
custom domain policy can be created.
7.2.1. Media Rules
Media Rules define RTP media packet parameters such as prioritizing encryption techniques and
packet encryption techniques. Together these media-related parameters define a strict profile that
is associated with other SIP-specific policies to determine how media packets matching these
criteria will be handled by the UC-Sec security product.
Create a custom Media Rule to set the Quality of Service and Media Anomaly Detection. The
sample configuration shows a custom Media Rule New-Low-Med created for the enterprise and
Windstream.
To create a custom Media Rule, navigate to UC-Sec Control Center Domain Policies 
Media Rules. With default-low-med selected, click Clone Rule as shown below.
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Enter a descriptive name for the new rule and click Finish.
When the RTP packets of a call are shuffled from Communication Manager to an IP Phone,
Avaya SBCE will interpret this as an anomaly and an alert will be created in the Incidents Log.
Disabling Media Anomaly Detection prevents the RTP Injection Attack alerts from being
created during an audio shuffle. To modify the rule, select the Media Anomaly tab and click
Edit. Uncheck Media Anomaly Detection and click Finish (not shown).
The following screen shows the New-Low-Med rule with Media Anomaly Detection disabled.
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On the Media QoS tab select the proper Quality of Service (QoS). Avaya SBCE can be
configured to mark the Differentiated Services Code Point (DSCP) in the IP Header with specific
values to support Quality of Services policies for the media. The following screen shows the
QoS values used for compliance testing.
7.2.2. Signaling Rules
Signaling Rules define the action to be taken (Allow, Block, Block with Response, etc.) for each
type of SIP-specific signaling request and response message. When SIP signaling packets are
received by the UC-Sec, they are parsed and “pattern-matched” against the particular signaling
criteria defined by these rules. Packets matching the criteria defined by the Signaling Rules are
tagged for further policy matching.
Clone and modify the default signaling rule to have the Avaya SBCE respond to SIP OPTION
requests and to set the Quality of Service. To clone a signaling rule, navigate to UC-Sec Control
Center Domain Policies  Signaling Rules. With the default rule chosen, click on Clone
Rule as shown below.
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Enter a descriptive name for the new rule and click Finish.
On the Requests tab, click on Add In Request Control to add a new Request Control to block
OPTIONS request from passing through the Avaya SBCE and return 200 OK as the response as
shown below.
On the Signaling QoS tab, select the proper Quality of Service (QoS). Avaya SBCE can be
configured to mark the Differentiated Services Code Point (DSCP) in the IP Header with specific
values to support Quality of Services policies for signaling. The following screen shows the QoS
values used for compliance testing.
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7.2.3. Application Rules
Application Rules define which types of SIP-based Unified Communications (UC) applications
the UC-Sec security device will protect: voice, video, and/or Instant Messaging (IM). In
addition, you can determine the maximum number of concurrent voice and video sessions the
network will process in order to prevent resource exhaustion.
Create an Application Rule to set the number of concurrent voice traffic. The sample
configuration cloned and modified the default application rule to increase the number of
Maximum Concurrent Session and Maximum Sessions Per Endpoint. To clone an
application rule, navigate to UC-Sec Control Center Domain Policies  Application Rules.
With the default rule chosen, click on Clone Rule as shown below.
Enter a descriptive name for the new rule and click Finish.
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Modify the rule by clicking the Edit button. Set the Maximum Concurrent Sessions and
Maximum Session Per Endpoint for the Voice application to a value high enough for the
amount of traffic the network is able process. Keep in mind Avaya SBCE takes 30 seconds for
sessions to be cleared after disconnect. The following screen shows the modified Application
Rule with the Maximum Concurrent Sessions and Maximum Session Per Endpoint set to
2000. In the sample configuration, Communication Manager was programmed to control the
concurrent sessions by setting the number of members in the trunk group (Section 5.8) to the
allotted amount. Therefore, the values in the Application Rule MaxVoiceSession were set high
enough to be considered non-blocking.
7.2.4. Endpoint Policy Group
The rules created within the Domain Policy section are assigned to an Endpoint Policy Group.
The Endpoint Policy Group is then applied to a Server Flow in Section 7.3.4. Create a separate
Endpoint Policy Group for the enterprise and the Windstream SIP Trunking service.
To create a new policy group, navigate to UC-Sec Control Center Domain Policies 
Endpoint Policy Groups and click on Add Group as shown below.
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The following screen shows Enterprise_DomPolicy created for the enterprise. Set the
Application, Media and Signaling rules to the ones previously created. Set the Border and
Time of Day rules to default and set the Security rule to default-low.
The following screen shows SIP Trunk_DomPolicy created for Windstream. Set the
Application, Media and Signaling rules to the ones previously created. Set the Border,
Signaling, and Time of Day rules to default and set the Security rule to default-high.
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7.3. Device Specific Settings
The Device Specific Settings feature allows aggregate system information to be viewed, and
various device-specific parameters to be managed to determine how a particular device will
function when deployed in the network. Specifically, it gives the ability to define and administer
various device-specific protection features such as Message Sequence Analysis (MSA)
functionality and protocol scrubber rules, end-point and session call flows, as well as the ability
to manage system logs and control security features.
7.3.1. Network Management
The Network Management screen is where the network interface settings are configured and
enabled. During the installation process of Avaya SBCE, certain network-specific information is
defined such as device IP address(es), public IP address(es), netmask, gateway, etc. to interface
the device to the network. It is this information that populates the various Network
Management tab displays, which can be edited as needed to optimize device performance and
network efficiency.
Navigate to UC-Sec Control Center Device Specific Settings  Network Management and
verify the IP addresses assigned to the interfaces and that the interfaces are enabled. The
following screen shows the private interface is assigned to A1 and the external interface is
assigned to B1.
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Enable the interfaces used to connect to the inside and outside networks on the Interface
Configuration tab. The following screen shows interface A1 and B1 are Enabled. To enable an
interface click it’s Toggle State button.
7.3.2. Signaling Interface
The Signaling Interface screen is where the SIP signaling ports are defined. Avaya SBCE will
listen for SIP requests on the defined ports. Create a Signaling Interface for both the inside and
outside IP interfaces.
To create a new Signaling Interface, navigate to UC-Sec Control Center Device Specific
Settings  Signaling Interface and click Add Signaling Interface.
The following screen shows the signaling interfaces created in the sample configuration with
TCP and UDP ports 5060 used for the inside and outside IP interfaces.
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7.3.3. Media Interface
The Media Interface screen is where the SIP media ports are defined. Avaya SBCE will listen
for SIP media on the defined ports. Create a SIP Media Interface for both the inside and outside
IP interfaces. The inside port range needs to match the UDP Port Min and UDP Port Max
fields in the Communication Manager IP network Region created in Section 5.6.
To create a new Media Interface, navigate to UC-Sec Control Center  Device Specific
Settings  Media Interface and click Add Media Interface.
The following screen shows the media interfaces created in the sample configuration for the
inside and outside IP interfaces. After the media interfaces are created, an application restart is
necessary before the changes will take effect.
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7.3.4. End Point Flows - Server Flow
When a packet is received by UC-Sec, the content of the packet (IP addresses, URIs, etc.) is used
to determine which flow it matches. Once the flow is determined, the flow points to a policy
which contains several rules concerning processing, privileges, authentication, routing, etc. Once
routing is applied and the destination endpoint is determined, the policies for this destination
endpoint are applied. The context is maintained, so as to be applied to future packets in the same
flow. The following screen illustrates the flow through Avaya SBCE to secure a SIP Trunk call.
Create a Server Flow for Session Manager and Windstream. To create a Server Flow, navigate
to UC-Sec Control Center  Device Specific Settings  End Point Flows. Select the Server
Flows tab and click Add Flow as shown below.
In the new window that appears, enter the following values. Use default values for all remaining
fields:


Flow Name:
Server Configuration:

Received Interface:
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Enter a descriptive name.
Select a Server Configuration created in Section 7.1.5 to
assign to the Flow.
Select the Signaling Interface the Server Configuration is
allowed to receive SIP messages from.
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
Signaling Interface:

Media Interface:


End Point Policy Group:
Routing Profile:

Topology Hiding Profile:
Select the Signaling Interface used to communicate with
the Server Configuration.
Select the Media Interface used to communicate with the
Server Configuration.
Select the policy assigned to the Server Configuration.
Select the profile the Server Configuration will use to route
SIP messages to.
Select the profile to apply toward the Server Configuration.
Click Finish to save and exit.
The following screen shows the Sever Flow for Windstream:
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The following screen shows the Sever Flow for Session Manager:
8. Windstream SIP Trunking Configuration
Windstream is responsible for the configuration of Windstream SIP Trunking. The customer
will need to provide the IP address used to reach the Avaya SBCE. Windstream will provide the
customer the necessary information to configure Communication Manager, Session Manager and
Avaya SBCE to connect to Windstream including:




IP address of the Windstream SIP proxy
Supported codecs
DID numbers
All IP addresses and port numbers used for signaling or media that will need access to the
enterprise network through any security devices.
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9. Verification and Troubleshooting
This section provides verification steps that may be performed in the field to verify that the
solution is configured properly. This section also provides a list of useful troubleshooting
commands that can be used to troubleshoot the solution.
9.1. Verification
The following steps may be used to verify the configuration:
1. Verify the call routing administration on Session Manager by logging in to System
Manager and executing the Call Routing Test. Expand Elements  Session Manager
 System Tools  Call Routing Test. Populate the field for the call parameters of
interest. For example, the following screen shows an example call routing test for an
outbound call to PSTN via Windstream. Under Routing Decisions, observe the call will
rout via the Avaya SBCE SIP Entity to Windstream. Scroll down to inspect the details of
the Routing Decision Process if desired (not shown).
2. Verify that endpoints at the enterprise site can place calls to the PSTN and that the call
remains active for more than 35 seconds. This time period is included to verify that
proper routing of the SIP messaging has satisfied SIP protocol timers.
3. Verify that endpoints at the enterprise site can receive calls from the PSTN and that the
call can remain active for more than 35 seconds.
4. Verify that the user on the PSTN can end an active call by hanging up.
5. Verify that an endpoint at the enterprise site can end an active call by hanging up.
Use the SAT interface on Communication Manager to verify status of SIP trunks. Specifically
use the status trunk n command to verify the active call has ended. Where n is the trunk group
number used for Windstream SIP Trunking defined in Section 5.8.
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Below is an example of an active call.
status trunk 1
TRUNK GROUP STATUS
Member
Port
Service State
Mtce Connected Ports
Busy
0001/001
0001/002
0001/003
0001/004
o
T00001
T00002
T00003
T00004
in-service/active
in-service/idle
in-service/idle
in-service/idle
no
no
no
no
S00000
Verify the port returns to in-service/idle after the call has ended.
status trunk 1
TRUNK GROUP STATUS
Member
Port
Service State
Mtce Connected Ports
Busy
0001/001
0001/002
0001/003
0001/004
T00001
T00002
T00003
T00004
in-service/idle
in-service/idle
in-service/idle
in-service/idle
no
no
no
no
9.2. Troubleshooting
1. Communication Manager:
 list trace station <extension number> - Traces calls to and from a specific
station.
 list trace tac <trunk access code number> - Trace calls over a specific trunk
group.
 status station <extension number> - Displays signaling and media information
for an active call on a specific station.
 status trunk <trunk number> - Displays trunk group information.
2. Session Manager:
 traceSM -x -uni - Session Manager command line tool for traffic analysis. Login
to the Session Manager management interface to run this command.
3. Avaya SBCE:
 Incidences - Displays alerts captured by the UC-Sec appliance.
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
Diagnostics - Allows for PING tests and displays application and protocol use.

Troubleshooting  Trace Settings - Configure and display call traces and
packet captures for the UC-Sec appliance.
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The packet capture file can be downloaded and viewed using a Network Protocol Analyzer:
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10. Conclusion
These Application Notes describe the configuration necessary to connect Avaya Session Border
Controller for Enterprise, Avaya Aura® Session Manager, and Avaya Aura® Communication
Manager Evolution Server to the Windstream SIP Trunking service. The Windstream SIP
Trunking service is a SIP-based Voice over IP solution for customers ranging from small
businesses to large enterprises. The Windstream SIP Trunking service provides businesses a
flexible, cost-saving alternative to traditional hardwired telephony trunks.
11. References
This section references the documentation relevant to these Application Notes. Additional
Avaya product documentation is available at http://support.avaya.com. Avaya SBCE product
documentation is available at http://www.sipera.com.
[1] Installing and Configuring Avaya Aura® System Platform, Release 6.0.3, February 2011.
[2] Administering Avaya Aura® System Platform, Release 6.0.3, February 2011.
[3] Administering Avaya Aura® Communication Manager, June2010, Document Number 03300509.
[4] Avaya Aura® Communication Manager Feature Description and Implementation, June 2010,
Document Number 555-245-205.
[5] Installing and Upgrading Avaya Aura® System Manager 6.1 GA Version, November 2010.
[6] Installing and Configuring Avaya Aura® Session Manager, April 2011, Document Number 03603473
[7] Administering Avaya Aura® Session Manager, November 2010, Document Number 03603324.
[8] Avaya 1600 Series IP Deskphones Administrator Guide Release 1.3.x, April 2010, Document
Number 16-601443.
[9] 4600 Series IP Telephone LAN Administrator Guide, July 2008, Document Number 555-233507.
[10] Avaya one-X Deskphone H.323 Administrator Guide, May 2011, Document Number 16300698.
[11] Avaya one-X Deskphone SIP Administrator Guide Release 6.1, December 2010, Document
Number 16-603838
[12] Administering Avaya one-X Communicator, July 2011
[13] Administrator Guide for Avaya Communication Manager, February 2007, Issue 3,
Document Number 03-300509.
[14] Feature Description and Implementation for Avaya Communication Manager, Issue 5,
Document Number 555-245-205
[15] UC-Sec Install Guide (102-5224-400v1.01)
[16] UC-Sec Administration Guide (010-5423-400v106)
[17] RFC 3261 SIP: Session Initiation Protocol, http://www.ietf.org/
[18] RFC 3515, The Session Initiation Protocol (SIP) Refer Method, http://www.ietf.org/
[19] RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals,
http://www.ietf.org/
[20] RFC 4244, An Extension to the Session Initiation Protocol (SIP) for Request History
Information, http://www.ietf.org/
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©2012 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at devconnect@avaya.com.
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