Avaya SM61 User Manual

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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.

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

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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

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

-

The following equipment and software were used for the reference configuration:

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5. Configure Communication Manager

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.)

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.

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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: 1 The 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 y to 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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Step

Description

6.

Trunk Group

Trunk group 1 was used for 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. Trunk group 1 was configured using the parameters highlighted below.

 Group Type: sip This field sets the type of the trunk group.  TAC: 101 Enter an valid value consistent with the Communication Manager dial

plan

 Member Assignment Method: auto Set to Auto.  Signaling Group: 1 This field is set to the signaling group shown in the previous

step.

 Number of Members: 50 This field represents the number of trunk group

members in the SIP trunk group. It determines how many simultaneous SIP calls can be supported by the configuration. Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. Thus, a call from a SIP telephone to another SIP telephone will use two SIP trunks. A call between a non-SIP telephone and a SIP telephone will only use one trunk.

display trunk-group 1 Page 1 of 21 TRUNK GROUP

Group Number: 1 Group Type: sip CDR Reports: y Group Name: to SM_21_31 COR: 1 TN: 1 TAC: 101 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Member Assignment Method: auto

Signaling Group: 1 Number of Members: 50

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Step

Description

Trunk Group – continued On Page 3:

 The Numbering Format field was set to unk-pvt. This field specifies the format

of the calling party number sent to the far-end.

 The default values may be retained for the other fields.

display trunk-group 1 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y

Numbering Format: unk-pvt UUI Treatment: service-provider

Replace Restricted Numbers? n Replace Unavailable Numbers? n

Modify Tandem Calling Number: no

Show ANSWERED BY on Display? y

7.

Private Numbering

Private Numbering defines the calling party number to be sent to the far-end. In the example shown below, all calls originating from a 5-digit extension beginning with 5 and routed across any trunk group will be sent as a 5 digit calling number. The calling party number is sent to the far-end in the SIP “From” header.

display private-numbering 0 Page 1 of 2 NUMBERING - PRIVATE FORMAT

Ext Ext Trk Private Total Len Code Grp(s) Prefix Len 5 5 5 Total Administered: 1 Maximum Entries: 540

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Step

Description

8.

Automatic Alternate Routing

Automatic Alternate Routing (AAR) was used to route calls either to Session Manager or to the Communication Manager at the other site. Use the change aar analysis command to create an entry in the AAR Digit Analysis Table. The example below shows numbers that begin with 75 and are 5 digits long use route pattern 1 (to Session Manager). Numbers that begin with 20000 or 65 and are 5 digits long use route pattern 7, which routes calls to Communication Manager at the other site via a SIP trunk (route pattern 8 was also used at times to route calls to Communication Manager at the other site via an ISDN-PRI trunk).

display aar analysis 2 Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 1

Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd 2 3 3 5 aar n 20000 5 5 7 aar n 23 5 5 8 aar n 531 5 5 1 unku n 532 5 5 1 unku n 59997 5 5 99 aar n 65 5 5 7 aar n

75 5 5 1 aar n

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Step

Description

9.

Route Pattern

Route pattern 1 was used for calls destined for the XMediusFAX fax server through Session Manager. Route patterns 7 and 8 (not shown) were used for calls destined for the other site in the reference configuration. Route pattern 1 was configured using the parameters highlighted below.

 Pattern Name: Any descriptive name.  Grp No: 1 This field is set to the trunk group number defined in Step 5.  FRL: 0 This field sets the Facility Restriction Level of the trunk. It must be set to

an appropriate level to allow authorized users to access the trunk. The level of 0 is the least restrictive.

change route-pattern 1 Page 1 of 3 Pattern Number: 1 Pattern Name: to SM_21_31 SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 1 0 0 n user 2: n user 3: n user 4: n user 5: n user 6: n user

BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR 0 1 2 M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest lev0-pvt none 2: y y y y y n n rest none 3: y y y y y n n rest none 4: y y y y y n n rest none 5: y y y y y n n rest none 6: y y y y y n n rest none

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6. Configure Session Manager

This section provides the procedures for configuring Session Manager (version 6.1) as provisioned at Site 2 in the reference configuration. Although not shown is this document, a similar Session Manager configuration would be required at Site 1 (using the appropriate version

6.0 screens). All provisioning for Session Manager is performed via the System Manager web interface.

The following sections assume that Session Manager and System Manager have been installed and that network connectivity exists between the two platforms.

The Session Manager server provides the network interface for all inbound and outbound SIP signaling and media transport to all provisioned SIP entities. During compliance testing, the IP address assigned to the Security Module interface is 10.64.21.31 as specified in Figure 1. The Session Manager server also has a separate network interface used for connectivity to System Manager for provisioning Session Manager. The IP address assigned to the Session Manager management interface is 10.64.21.30.

The procedures described in this section include configurations in the following areas:

 SIP Domains – SIP Domains are the domains for which Session Manager is authoritative in

routing SIP calls. In other words, for calls to such domains, Session Manager applies Network Routing Policies to route those calls to SIP Entities. For calls to other domains, Session Manager routes those calls to another SIP proxy (either a pre-defined default SIP proxy or one discovered through DNS).

 Locations – Locations define the physical and/or logical locations in which SIP Entities

reside. Call Admission Control (CAC) / bandwidth management may be administered for each location to limit the number of calls to and from a particular Location.

 Adaptations – Adaptations are used to apply any necessary protocol adaptations, e.g.,

modify SIP headers, and apply any necessary digit conversions for the purpose of interworking with specific SIP Entities.

 SIP Entities – SIP Entities represent SIP network elements such as Session Manager

instances, Communication Manager systems, Session Border Controllers, SIP gateways, SIP trunks, and other SIP network devices.

 Entity Links – Entity Links define the SIP trunk/link parameters, e.g., ports, protocol

(UDP/TCP/TLS), and trust relationship, between Session Manager instances and other SIP Entities.

 Time Ranges – Time Ranges specify customizable time periods, e.g., Monday through

Friday from 9AM to 5:59PM, Monday through Friday 6PM to 8:59AM, all day Saturday and Sunday, etc. A Network Routing Policy may be associated with one or more Time Ranges during which the Network Routing Policy is in effect.

 Routing Policies – Routing Policies are used in conjunction with a Dial Patterns to

specify a SIP Entity that a call should be routed to.

 Dial Patterns – A Dial Pattern specifies a set of criteria and a set of Network Routing

Policies for routing calls that match the criteria. The criteria include the called party number and SIP domain in the Request-URI, and the Location from which the call originated. For