Avaya Sagemcom XMediusFAX s Supplement

MJH; Reviewed:
SPOC 8/4/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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Sagemcom_SM61
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Sagemcom XMediusFAX
Service Provider Edition with Avaya Aura® Session
Manager and Avaya Aura® Communication Manager
- Issue 1.0
Abstract
These Application Notes describe the procedures for configuring Sagemcom XMediusFAX
Service Provider (SP) Edition with Avaya Aura® Session Manager and Avaya Aura®
Communication Manager.
XMediusFAX is a software based fax server that sends and receives fax calls over an IP
network. In the configuration tested, XMediusFAX interoperates with Avaya Aura® Session
Manager and Avaya Aura® Communication Manager to send/receive faxes using SIP trunks
and the T.38 fax protocol between XMediusFAX and the Avaya SIP infrastructure.
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.

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1. Introduction
These Application Notes describe the procedures for configuring Sagemcom XMediusFAX
Service Provider (SP) Edition with Avaya Aura® Session Manager and Avaya Aura®
Communication Manager using SIP trunks.
XMediusFAX is a software based fax server that sends and receives fax calls over an IP network.
In the configuration tested, XMediusFAX interoperates with the Session Manager and
Communication Manager to send/receive faxes using SIP trunks and the T.38 protocol between
XMediusFAX and the Avaya SIP infrastructure. The compliance testing focused on fax calls to
and from the XMediusFAX fax server using various page lengths, resolutions, and fax data
speeds.
2. General Test Approach and Test Results
This section describes the general test approach used to verify the interoperability of Sagemcom
XMediusFAX SP Edition with the Avaya SIP infrastructure (Session Manager and
Communication Manager). This section also covers the test results.
2.1. Interoperability Compliance Testing
The general test approach was to make intra-site and inter-site fax calls to and from the
XMediusFAX fax server. The compliance tested configuration contained two sites. Site 2 served
as the main enterprise site and Site 1 served as a simulated PSTN or a remote enterprise site.
Inter-site calls and simulated PSTN calls were made using SIP trunks and ISDN-PRI trunks
between the sites. By using two Communication Managers and two port networks at Site 1, fax
calls across multiple TDM/IP hops were able to be tested. Faxes were sent with various page
lengths, resolutions, and at various fax data speeds. For capacity testing, 100 2-page faxes were
continuously sent between the two XMediusFAX fax servers. Serviceability testing included
verifying proper operation/recovery from network outages, unavailable resources, and
Communication Manager and XMediusFAX restarts. Fax calls were also tested with different
Avaya Media Gateway media resources to process the fax data. This included the TN2302
MedPro circuit pack, the TN2602 MedPro circuit pack in the Avaya G650 Media Gateway; and
the integrated VoIP engine of the Avaya G450 Media Gateway.
2.2. Test Results
XMediusFAX successfully passed compliance testing.
2.3. Support
For technical support on XMediusFAX, contact Sagemcom at:
Web: http://xmediusfax.sagemcom.com/support/
Phone: [phone]
Email: [e-mail]

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3. Reference Configuration
Figure 1 illustrates the reference configuration used during testing. In the reference
configuration, the two sites are connected via a direct SIP trunk and an ISDN-PRI trunk. Faxes
were sent between the two sites using either of these two trunks, as dictated by each individual
test case.
Figure 1: XMediusFAX with Session Manager and Communication Manager
At Site 1 consists of the following equipment:
An Avaya S8800 Server running Avaya Aura® Communication Manager with two
Avaya G650 Media Gateways. Each media gateway is configured as a separate port
network in separate IP network regions. The media resources required are provided by
the IP Media Processor (MedPro) circuit packs. Two versions of the IP MedPro circuit
pack were tested in the configuration: the TN2302AP and the TN2602AP.
An Avaya S8800 Server running Avaya Aura® System Manager. System Manager
provides management functions for Session Manager.
An Avaya S8800 Server running Avaya Aura® Session Manager.
XMediusFAX running on a Windows 2008 R2 Enterprise Server (SP1).
An analog fax machine.
Various Avaya IP endpoints (not all shown).
At Site 2 consists of the following equipment:
An Avaya S8300D Server running Avaya Aura® Communication Manager in an Avaya
G450 Media Gateway. The signaling and media resources needed to support SIP trunks
are integrated directly on the media gateway processor.

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A Dell™ PowerEdge™ R610 Server running Avaya Aura® System Manager. System
Manager provides management functions for Session Manager.
An HP ProLiant DL360 G7 Server running Avaya Aura® Session Manager.
XMediusFAX running on a Windows 2008 R2 Enterprise Server (SP1).
An analog fax machine
Various Avaya IP endpoints (not all shown).
Although the IP endpoints (H.323 and SIP telephones) are not involved in the faxing operations,
they are present at both sites to verify that VoIP telephone calls are not affected by the FoIP
faxing operations and vice versa.
Outbound fax calls originating from the XMediusFAX fax server are sent to Session Manager
first, and then from Session Manager to Communication Manager via SIP trunks. Based on the
dialed digits, Communication Manager will either direct the calls to the local fax machine, or to
the other site via an ISDN-PRI or SIP trunk. Inbound fax calls terminating to the XMediusFAX
fax server are sent from the local fax machine or from the remote site are received by
Communication Manager. The calls are then directed to Session Manager for onward routing to
the XMediusFAX fax server via SIP trunks.

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4. Equipment and Software Validated
The following equipment and software were used for the reference configuration:
Equipment
Software
Site 1
Avaya S8800 Server with a Avaya G650
Media Gateways:
-2 CLANs –TN799DP
-2 IP MedPros –TN2302AP
-2 IP MedPros –TN2602AP
Avaya Aura® Communication Manager 6.0.1,
R016x.00.1.510.1, Patch 19009 :
-HW01, FW038
-HW20, FW120
-HW02, FW57
Avaya S8800 Server
Avaya Aura® System Manager: 6.0.0 (Build No. –
6.0.0.0.688-3.0.7.2)
(Avaya Aura® System Platform: 6.0.2.1.5)
Avaya S8800 Server
Avaya Aura® Session Manager 6.0.2.0.602004
XMediusFAX fax server (Windows 2008
R2 Enterprise Server, SP1)
6.5.5 with patch XMFSP_6.5.5.213
Fax Machine
-
Various Avaya SIP and H.323 endpoints
-
Site 2
Avaya S8300D Server with a Avaya G450
Media Gateway
Avaya Aura® Communication Manager 6.0.1,
R016x.00.1.510.1, Patch 19009
(Avaya Aura® System Platform: 6.0.3.0.3)
Dell™ PowerEdge™ R610 Server
Avaya Aura® System Manager: 6.1.0 (Build No. –
6.1.0.0.7345-6.1.5.106), Software Update Revision
No : 6.1.6.1.1087
(Avaya Aura® System Platform: 6.0.3.0.3)
HP ProLiant DL360 G7 Server
Avaya Aura® Session Manager 6.1.2.0.612004
XMediusFAX fax server (Windows 2008
R2 Enterprise Server, SP1)
6.5.5 with patch XMFSP_6.5.5.213
Fax Machine
-
Various Avaya SIP and H.323 endpoints
-

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5. Configure Communication Manager
This section describes the Communication Manager configuration at Site 2 to support the
network shown in Figure 1. Although not shown is this document, a similar Communication
Manager configuration would be required at Site 1.
The configuration of Communication Manager was performed using the System Access
Terminal (SAT). After the completion of the configuration, perform a save translation
command to make the changes permanent.
Step
Description
1.
License
Use the display system-parameters customer-options command to verify that the
Communication Manager license has proper permissions for features illustrated in
these Application Notes. Navigate to Page 2, and verify that there is sufficient
remaining capacity for SIP trunks by comparing the Maximum Administered SIP
Trunks field value with the corresponding value in the USED column.
The license file installed on the system controls the maximum permitted. If there is
insufficient capacity, contact an authorized Avaya sales representative to make the
appropriate changes.
display system-parameters customer-options Page 2 of 11
OPTIONAL FEATURES
IP PORT CAPACITIES USED
Maximum Administered H.323 Trunks: 12000 32
Maximum Concurrently Registered IP Stations: 18000 15
Maximum Administered Remote Office Trunks: 12000 0
Maximum Concurrently Registered Remote Office Stations: 18000 0
Maximum Concurrently Registered IP eCons: 414 0
Max Concur Registered Unauthenticated H.323 Stations: 100 0
Maximum Video Capable Stations: 18000 0
Maximum Video Capable IP Softphones: 18000 1
Maximum Administered SIP Trunks: 24000 170
Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0
Maximum Number of DS1 Boards with Echo Cancellation: 522 0
Maximum TN2501 VAL Boards: 128 0
Maximum Media Gateway VAL Sources: 250 1
Maximum TN2602 Boards with 80 VoIP Channels: 128 0
Maximum TN2602 Boards with 320 VoIP Channels: 128 0
Maximum Number of Expanded Meet-me Conference Ports: 300 0
(NOTE: You must logoff & login to effect the permission changes.)

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Step
Description
2.
IP network region
Use the display ip-network-region command to view the network region settings.
The values shown below are the values used during compliance testing.
Authoritative Domain: avaya.com This field was configured to match the
domain name configured on Session Manager. The domain will appear in the
“From” header of SIP messages originating from this IP region.
Name: Any descriptive name may be used (if desired).
Intra-region IP-IP Direct Audio: yes
Inter-region IP-IP Direct Audio: yes
By default, IP-IP direct audio (media shuffling) is enabled to allow audio traffic to
be sent directly between IP endpoints without using media resources in the Avaya
Media Gateway. Shuffling can be further restricted at the trunk level on the
Signaling Group form.
Codec Set: 1The codec set contains the list of codecs available for calls within
this IP network region.
Display ip-network-region 1 Page 1 of 20
IP NETWORK REGION
Region: 1
Location: Authoritative Domain: avaya.com
Name: FAX testing
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 1 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

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Step
Description
3.
Codecs
IP codec set 1 was used during compliance testing. Multiple codecs can be listed in
priority order to allow the codec used by a specific call to be negotiated during call
establishment. The example below shows the values used during compliance testing.
display ip-codec-set 1 Page 1 of 2
IP Codec Set
Codec Set: 1
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.711MU n 2 20
2:
On Page 2, set the FAX Mode field to t.38-standard. The Modem Mode field should
be set to off.
Leave the FAX Redundancy setting at its default value of 0.A packet redundancy
level can be assigned to improve packet delivery and robustness of FAX transport over
the network (with increased bandwidth as trade-off). Avaya uses IETF RFC-2198 and
ITU-T T.38 specifications as redundancy standard. With this standard, each Fax over
IP packet is sent with additional (redundant) 0 to 3 previous fax packets based on the
redundancy setting. A setting of 0 (no redundancy) is suited for networks where packet
loss is not a problem.
display ip-codec-set 1 Page 2 of 2
IP Codec Set
Allow Direct-IP Multimedia? y
Maximum Call Rate for Direct-IP Multimedia: 2048:Kbits
Maximum Call Rate for Priority Direct-IP Multimedia: 2048:Kbits
Mode Redundancy
FAX t.38-standard 0
Modem off 0
TDD/TTY US 3
Clear-channel n 0

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Step
Description
4.
Node Names
Use the change node-names ip command to create a node name for the IP address of
Session Manager. Enter a descriptive name in the Name column and the IP address
assigned to Session Manager in the IP address column.
change node-names ip Page 1 of 2
IP NODE NAMES
Name IP Address
AES_21_46 10.64.21.46
CM_20_40 10.64.20.40
CM_22_12_CLAN1A 10.64.22.16
CM_22_12_CLAN2A 10.64.22.19
IPO_21_64 10.64.21.64
SM_20_31 10.64.20.31
SM_21_31 10.64.21.31
default 0.0.0.0
msgserver 10.64.21.41
procr 10.64.21.41
procr6 ::

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Step
Description
5.
Signaling Group
Signaling group 1 was used for the signaling group associated with the SIP trunk group
between Communication Manager and Session Manager. The signaling groups and
trunk groups between the two sites of the reference configuration is assumed to already
be in place and not described in this document. Signaling group 1 was configured
using the parameters highlighted below.
Near-end Node Name: procr This node name maps to the IP address of the Avaya
S8300D Server. Node names are defined using the change node-names ip
command.
Far-end Node Name: SM_21_31 This node name maps to the IP address of
Session Manager.
Far-end Network Region: 1 This defines the IP network region which contains
Session Manager.
Far-end Domain: avaya.com This domain is sent in the “To” header of SIP
messages of calls using this signaling group.
Direct IP-IP Audio Connections: y This field must be set to yto enable media
shuffling on the SIP trunk.
display signaling-group 1
SIGNALING GROUP
Group Number: 1 Group Type: sip
IMS Enabled? n Transport Method: tls
Q-SIP? n SIP Enabled LSP? n
IP Video? y Priority Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM
Near-end Node Name: procr Far-end Node Name: SM_21_31
Near-end Listen Port: 5061 Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain: avaya.com
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6