Avaya Sagemcom XMediusFAX s Suplemento

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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: (888) 766-1668

Email: xmediusfax.support.americas@sagemcom.com

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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

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

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

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

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

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

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

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