Abstract. Avaya Solution & Interoperability Test Lab

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Avaya Solution & Interoperability Test Lab

Application Notes for Configuring the AudioCodes Mediant

1000 VoIP Media Gateway with Avaya SIP Enablement

Services and Avaya Communication Manager - Issue 1.0

Abstract

These Application Notes describe the procedure for configuring the AudioCodes Mediant 1000 VoIP Media Gateway to interoperate with Avaya SIP Enablement Services and Avaya Communication Manager.

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1. Introduction

These Application Notes describe the procedure for configuring the AudioCodes Mediant 1000 VoIP Media Gateway to interoperate with Avaya SIP Enablement Services (SES) and Avaya

Communication Manager.

The AudioCodes Mediant 1000 VoIP Media Gateway serves as a gateway between TDM and IP networks. The Mediant 1000 supports multiple hardware interfaces and control protocols. Capacity can be scaled upward by adding additional interface modules. The compliance test configured the Mediant 1000 as a SIP to ISDN-PRI gateway connecting two enterprise sites.

1.1. Configuration

Figure 1 illustrates the configuration used in these Application Notes. In the sample configuration,

two sites are connected via an IP network. Located at Site 1 is an Avaya SES and an Avaya S8300 Media Server running Avaya Communication Manager in an Avaya G350 Media Gateway.

Endpoints include an Avaya 4600 Series IP Telephone with SIP firmware, an Avaya 4600 Series IP Telephone with H.323 firmware, an Avaya 6408D Digital Telephone, and a fax machine. There is no direct connection to the PSTN at this site.

Located at Site 2 is an AudioCodes Mediant 1000 VoIP Media Gateway connected to an Avaya DEFINITY Server R via an PRI trunk. The Avaya DEFINITY Server R in turn has an ISDN-PRI connection to the PSTN. Site 2 provides PSTN access for both sites. All outbound calls from any phone at Site 1 will be routed across the SIP connection starting at Avaya Communication Manager at Site 1 through Avaya SES and terminating at the Mediant 1000 at Site 2. The Mediant 1000 routes the call from its SIP interface to its ISDN-PRI interface which is connected to the Avaya DEFINITY Server R. The Avaya DEFINITY Server R then routes the call to the PSTN. Inbound calls to Site 1 follow the same path in the reverse order, entering the network at the Avaya DEFINITY Server R at Site 2.

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2. Equipment and Software Validated

The following equipment and software/firmware were used for the sample configuration provided:

Equipment Software/Firmware

Avaya S8300 Media Server with Avaya G350 Media Gateway

Avaya Communication Manager 3.1.2 (R013x.01.2.635.0)

This is a post-GA build needed to support T.38 fax with the Mediant

1000. Avaya SIP Enablement Services (SES) 3.1 (build 18)

Avaya DEFINITY Server R R011r.03.1.535.0

Avaya 4620SW IP Telephones SIP version 2.2.2

Avaya 4620 IP Telephones H.323 version 2.3

Avaya 6408D Digital Telephone -

Analog Telephones -

Analog Fax Machines -

Windows PCs Windows XP Professional

AudioCodes Mediant 1000 VoIP Media Gateway

5.00A.011.008

3. Configure Avaya Communication Manager

The following configuration of Avaya Communication Manager was performed using the System Access Terminal (SAT). After the completion of the configuration in this section, perform a save

translation command to make the changes permanent.

3.1. Configure SIP Related Components at Site 1

The communication between Avaya Communication Manager and Avaya SES at Site 1 is via a SIP trunk group. All SIP signaling for calls between Avaya Communication Manager and the Mediant 1000 passes through Avaya SES via this trunk group. This section describes the steps for

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Step Description

1. Use the display system-parameters customer-options command to verify that

sufficient SIP trunk capacity exists. On Page 2, verify that the number of SIP trunks supported by the system is sufficient for the number of SIP trunks needed. 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.

The license file installed on the system controls the maximum permitted. If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales representative to make the appropriate changes.

2. On Page 4, verify that the features shown in bold in the example below are enabled.

display system-parameters customer-options Page 2 of 10 OPTIONAL FEATURES

IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: 100 10 Maximum Concurrently Registered IP Stations: 20 0 Maximum Administered Remote Office Trunks: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP eCons: 0 0 Max Concur Registered Unauthenticated H.323 Stations: 0 0 Maximum Video Capable H.323 Stations: 0 0 Maximum Video Capable IP Softphones: 0 0 Maximum Administered SIP Trunks: 100 24 Maximum Number of DS1 Boards with Echo Cancellation: 0 0 Maximum TN2501 VAL Boards: 0 0 Maximum G250/G350/G700 VAL Sources: 5 1 Maximum TN2602 Boards with 80 VoIP Channels: 0 0 Maximum TN2602 Boards with 320 VoIP Channels: 0 0 Maximum Number of Expanded Meet-me Conference Ports: 10 0

(NOTE: You must logoff & login to effect the permission changes.)

Emergency Access to Attendant? y IP Stations? y Enable 'dadmin' Login? y Internet Protocol (IP) PNC? n Enhanced Conferencing? y ISDN Feature Plus? n Enhanced EC500? y ISDN Network Call Redirection? n Enterprise Survivable Server? n ISDN-BRI Trunks? n Enterprise Wide Licensing? n ISDN-PRI? y ESS Administration? n Local Survivable Processor? n Extended Cvg/Fwd Admin? n Malicious Call Trace? n External Device Alarm Admin? n Media Encryption Over IP? n Five Port Networks Max Per MCC? n Mode Code for Centralized Voice Mail? n Flexible Billing? n

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3. On Page 5, verify that the features shown in bold in the example below are enabled.

4. Use the change node-name ip command to assign the node name and IP address for

the Avaya SES at Site 1. In this case, SES and 50.1.1.50 are being used, respectively. The node name SES will be used throughout the other configuration forms of Avaya Communication Manager. In this example, procr and 50.1.1.10 are the name and IP address assigned to the Avaya S8300 Media Server.

Multinational Locations? n Station and Trunk MSP? n Multiple Level Precedence & Preemption? n Station as Virtual Extension? n Multiple Locations? n

System Management Data Transfer? n Personal Station Access (PSA)? n Tenant Partitioning? n Posted Messages? n Terminal Trans. Init. (TTI)? n PNC Duplication? n Time of Day Routing? n Port Network Support? n Uniform Dialing Plan? n Usage Allocation Enhancements? y Processor and System MSP? n TN2501 VAL Maximum Capacity? y Private Networking? y

Processor Ethernet? y Wideband Switching? n Wireless? n Remote Office? n

Restrict Call Forward Off Net? y Secondary Data Module? y

change node-names ip Page 1 of 1 IP NODE NAMES

Name IP Address Name IP Address

SES 50 .1 .1 .50 . . .

default 0 .0 .0 .0 . . .

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5. Use the change ip-network-region n command, where n is the number of the region

to be changed, to define the connectivity settings for all VoIP resources and IP endpoints within the region. Select an IP network region that will contain the Avaya SES server. The association between this IP network region and the Avaya SES server will be done on the Signaling Group form as shown in Step 8. In the case of the compliance test, the same IP network region that contains the Avaya S8300 Media Server and Avaya IP Telephones was selected to contain the Avaya SES server. By default, the Media Server and IP telephones are in IP Network Region 1.

On the IP Network Region form:

The Authoritative Domain field is configured to match the domain name configured on Avaya SES. In this configuration, the domain name is

devcon.com. This name will appear in the “From” header of SIP messages

originating from this IP region.

By default, IP-IP Direct Audio (shuffling) is enabled to allow audio traffic to be sent directly between IP endpoints without using media resources in the Avaya G350 Media Gateway. This is true for both intra-region and inter-region IP-IP Direct Audio. Shuffling can be further restricted at the trunk level on the

Signaling Group form.

The Codec Set is set to the number of the IP codec set to be used for calls within this IP network region. If different IP network regions are used for the Avaya S8300 Media Server and the Avaya SES server, then Page 3 of each IP

Network Region form must be used to specify the codec set for inter-region

communications.

The default values can be used for all other fields.

change ip-network-region 1 Page 1 of 19 IP NETWORK REGION

Region: 1

Location: 1 Authoritative Domain: devcon.com Name:

MEDIA PARAMETERS

Codec Set: 1 Inter-region IP-IP Direct Audio: yes Intra-region IP-IP Direct Audio: yes

UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 3027

DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 34 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y 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

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6. Use the change ip-codec-set n command, where n is the codec set value specified in

Step 5, to enter the supported audio codecs for calls routed to Avaya SES. Multiple codecs can be listed in priority order to allow the codec to be negotiated during call establishment. The list should include the codecs the enterprise wishes to support within the normal trade-off of bandwidth versus voice quality. The example below shows the values used in the compliance test.

7. On Page 2, the FAX Mode field must be set to t.38-standard to support the fax machine. The Modem field must be set to off. The screen below shows the setting used for the fax testing.

change 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: G.729AB n 2 20 3:

change ip-codec-set 1 Page 2 of 2 IP Codec Set

Allow Direct-IP Multimedia? n

Mode Redundancy FAX t.38-standard 0

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8. Use the add signaling group n command, where n is the number of an unused

signaling group, to create the SIP signaling group as follows: Set the Group Type field to sip.

The Transport Method field will default to tls (Transport Layer Security). TLS is the only link protocol that is supported for communication between Avaya SES and Avaya Communication Manager.

Specify the Avaya S8300 Media Server (node name procr) and the Avaya SES Server (node name SES) as the two ends of the signaling group in the

Near-end Node Name and the Far-Near-end Node Name fields, respectively. These field

values are taken from the IP Node Names form shown in Step 4. For alternative configurations that use a C-LAN board, the near (local) end of the SIP signaling group will be the C-LAN board instead of the Media Server. Ensure that the recommended TLS port value of 5061 is configured in the

Near-end Listen Port and the Far-end Listen Port fields.

In the Far-end Network Region field, enter the IP network region value assigned in the IP Network Region form in Step 5. This defines which IP network region contains the Avaya SES server. If the Far-end Network

Region field is different from the near-end network region, the preferred codec

will be selected from the IP codec set assigned for the inter-region connectivity for the pair of network regions.

Enter the domain name of Avaya SES in the Far-end Domain field. In this configuration, the domain name is devcon.com. This domain is specified in the Uniform Resource Identifier (URI) of the SIP “To” header in the INVITE message.

The Direct IP-IP Audio Connections field is set to y.

The DTMF over IP field must be set to the default value of rtp-payload for a SIP trunk. This value enables Avaya Communication Manager to send DTMF transmissions using RFC 2833.

The default values for the other fields may be used.

add signaling-group 1 Page 1 of 1 SIGNALING GROUP

Group Number: 1 Group Type: sip

Transport Method: tls

Near-end Node Name: procr Far-end Node Name: SES Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: 1 Far-end Domain: devcon.com

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9. Add a SIP trunk group by using the add trunk-group n command, where n is the number of an unused trunk group. For the compliance test, trunk group number 1 was chosen.

On Page 1, set the fields to the following values: Set the Group Type field to sip.

Choose a descriptive Group Name.

Specify an available trunk access code (TAC) that is consistent with the existing dial plan.

Set the Service Type field to tie.

Specify the signaling group associated with this trunk group in the Signaling

Group field as previously specified in Step 8.

Specify the Number of Members supported by this SIP trunk group. As mentioned earlier, 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. The default values may be retained for the other fields.

10. On Page 3:

Verify the Numbering Format field is set to public. 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.

add trunk-group 1 Page 1 of 21 TRUNK GROUP

Group Number: 1 Group Type: sip CDR Reports: y Group Name: To SES 50.1.1.50 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

Signaling Group: 1 Number of Members: 24

add trunk-group 1 Page 3 of 21 TRUNK FEATURES

ACA Assignment? n Measured: none

Maintenance Tests? y

Numbering Format: public

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11. Use the change public-unknown-numbering 0 command to define the full calling party number to be sent to the far-end. Add an entry for the trunk group defined in Step 9. In the example shown below, all calls originating from a 5-digit extension beginning with 4 and routed across trunk group 1 will be sent as a 5 digit calling number. This calling party number will be sent to the far-end in the SIP “From” header.

12. Create a route pattern that will use the SIP trunk that connects to Avaya SES. The compliance test used Automatic Route Selection (ARS) to define route pattern 1 as the route for all outbound calls. For more information on ARS see [1] and [2].

To create a route pattern, use the change route-pattern n command, where n is the number of an unused route pattern. Enter a descriptive name for the Pattern Name field. Set the Grp No field to the trunk group number created for the SIP trunk. 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. The default values may be retained for all other fields.

change public-unknown-numbering 0 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT

ota

Ext Ext Trk CPN CPN Ext Ext Trk CPN CPN

T l Total

Len Code Grp(s) Prefix Len Len Code Grp(s) Prefix Len

5 4 1 5

change route-pattern 1 Page 1 of 3 Pattern Number: 3 Pattern Name: SIP

SCCAN? n Secure SIP? n

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13. Use the change locations command to assign the default SIP route pattern to the location. In the compliance test, all SIP endpoints at Site 1 are part of a single location defined in Avaya Communication Manager. This location uses the default name of

Main and is shown in the example below. Enter the route pattern number from the

previous step in the Proxy Sel. Rte. Pat. field. The default values may be retained for all other fields.

14. To map a DID number to a station at Site 1, use the change inc-call-handling-trmt

trunk-group n command, where n is the SIP trunk group number connected to Site 2

defined in Step 9. All inbound calls arrive at Site 1 via Site 2 which has PSTN access. The example below shows two incoming 10-digit numbers being deleted and replaced with the extension number of the desired station.

change locations Page 1 of 4 LOCATIONS

ARS Prefix 1 Required For 10-Digit NANP Calls? y

Loc. Name Timezone Rule NPA ARS Attd Pre- Proxy Sel. No. Offset FAC FAC fix Rte. Pat. 1: Main + 00:00 0 1

2: 3:

change inc-call-handling-trmt trunk-group 1 Page 1 of 3 INCOMING CALL HANDLING TREATMENT

Service/ Called Called Del Insert Feature Len Number

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3.2. Configure ISDN-PRI Trunk Related Components at Site 2

1. Use the display system-parameters customer-options command to verify that the

ISDN-PRI feature is enabled on Page 3. If the feature is not enabled, contact an authorized Avaya sales representative to make the appropriate changes.

2. On Page 4, verify that Private Networking is enabled.

display system-parameters customer-options Page 3 of 10 SPE B OPTIONAL FEATURES

Emergency Access to Attendant? y ISDN Feature Plus? y Enable 'dadmin' Login? n ISDN Network Call Redirection? y Enhanced Conferencing? n ISDN-BRI Trunks? y Enhanced EC500? n ISDN-PRI? y Extended Cvg/Fwd Admin? y Local Spare Processor? n External Device Alarm Admin? y Malicious Call Trace? n Five Port Networks Max Per MCC? n Media Encryption Over IP? n Flexible Billing? n Mode Code for Centralized Voice Mail? n Forced Entry of Account Codes? n

Global Call Classification? n Multifrequency Signaling? y Hospitality (Basic)? y Multimedia Appl. Server Interface (MASI)? y Hospitality (G3V3 Enhancements)? y Multimedia Call Handling (Basic)? y IP Trunks? y Multimedia Call Handling (Enhanced)? y Multiple Locations? n IP Attendant Consoles? n Personal Station Access (PSA)? y IP Stations? y Posted Messages? n

display system-parameters customer-options Page 4 of 10 SPE B OPTIONAL FEATURES

PNC Duplication? n Tenant Partitioning? n Port Network Support? y Terminal Trans. Init. (TTI)? y Time of Day Routing? n Processor and System MSP? y Uniform Dialing Plan? y Private Networking? y Usage Allocation Enhancements? y TN2501 VAL Maximum Capacity? y Remote Office? y Wideband Switching? y Restrict Call Forward Off Net? y Wireless? n Secondary Data Module? y

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3. An ISDN-PRI trunk requires the use of a DS1 circuit pack. To configure the DS1 circuit pack, use the add ds1 n command where n is the location in the chassis of the DS1 circuit pack to be used. In the example below, the location is 1c07. The Name field can be any descriptive name. All other fields in bold in the example below should be set to the value shown. The combination of Country Protocol (1) and Protocol

Version (b) determine which version of ISDN-PRI will be used, specifically AT&T

5ESS. Default values may be retained for all other fields.

4. Use the add signaling-group n command, where n is the number of an unused

signaling group to be added. Set the fields in bold to the values shown below. The

Primary D-Channel field is set to the 24th channel of the DS1 board in slot 1c07. This board was added to the configuration in the previous step. The Trunk Group for

Channel Selection field will be populated at a later step after the trunk group has been

created.

add ds1 1c07 Page 1 of 2 SPE B DS1 CIRCUIT PACK

Location: 01C07 Name: Gw ds1 Bit Rate: 1.544 Line Coding: b8zs Line Compensation: 1 Framing Mode: esf Signaling Mode: isdn-pri

Connect: pbx Interface: network TN-C7 Long Timers? n Country Protocol: 1 Interworking Message: PROGress Protocol Version: a Interface Companding: mulaw CRC? n Idle Code: 11111111

DCP/Analog Bearer Capability: 3.1kHz

Slip Detection? n Near-end CSU Type: other Block Progress Indicator? n

add signaling-group 12 Page 1 of 5 SPE B SIGNALING GROUP

Group Number: 12 Group Type: isdn-pri

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5. Use the add trunk-group n command, where n is the number of an unused trunk

group, to be added. Set the fields in bold to the values shown below. The Group

Name can be any descriptive name. The TAC must be chosen to be consistent with

the existing dial plan.

6. On Page 2, set the fields in bold to the values shown below.

add trunk-group 12 Page 1 of 22 SPE B TRUNK GROUP

Group Number: 12 Group Type: isdn CDR Reports: y Group Name: Tim PRI 1 COR: 1 TN: 1 TAC: 112 Direction: two-way Outgoing Display? n Carrier Medium: PRI/BRI Dial Access? n Busy Threshold: 255 Night Service:

Queue Length: 0

Service Type: tie Auth Code? n TestCall ITC: rest

Far End Test Line No: TestCall BCC: 4

TRUNK PARAMETERS

Codeset to Send Display: 6 Codeset to Send National IEs: 6 Max Message Size to Send: 260 Charge Advice: none

Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc Trunk Hunt: cyclical QSIG Value-Added? n Digital Loss Group: 13 Calling Number - Delete: Insert: Numbering Format: Bit Rate: 1200 Synchronization: async Duplex: full Disconnect Supervision - In? y Out? n

Answer Supervision Timeout: 0

add trunk-group 12 Page 2 of 22 SPE B TRUNK FEATURES

ACA Assignment? n Measured: internal Wideband Support? n Internal Alert? n Maintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: y Send Calling Number: y Used for DCS? n

Suppress # Outpulsing? n Numbering Format: public

Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Send Connected Number: n Network Call Redirection: none

Send UUI IE? y Send UCID? n

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7. On Page 3, define the incoming call handling treatment for calls coming from the Mediant 1000 on this trunk. The entry in bold below specifies that all incoming calls of 11 digits in length will have *9 inserted at the beginning of the dial string. *9 is the feature access code for Automatic Route Selection (ARS) on the Avaya DEFINITY Server R. By adding *9 to the dial string, the call will be directed to the ARS tables which will select the route pattern shown in Step 10.

8. On Page 6, enter the group members. For each DS1 port to be added as a member of the trunk group, enter the port number in the Port field and the corresponding signaling group for that port in the Sig Grp field. The Code field is filled in automatically. In the compliance test, each of the 23 bearer channels of the DS1 board added in Step 3 were added to this group. Only the first 15 members are shown below. The signaling channel for each of these ports is the signaling channel added in Step 4.

add trunk-group 12 Page 3 of 22 SPE B INCOMING CALL HANDLING TREATMENT

Service/ Called Called Del Insert Per Call Night Feature Len Number CPN/BN Serv

tie 11 *9

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9. Use the change signaling-group 3 command to return to the Signaling Group form

shown in Step 4. Set the Trunk Group for Channel Selection field to the number of the trunk group created in Step 5.

10. The compliance testing used Automatic Route Selection (ARS) to define route pattern 12 as the route for all outbound calls to the PSTN on the Avaya DEFINITY Server R. For more information on ARS see [1] and [2].

The example below shows the route pattern used in the compliance test for all

outbound traffic. The Pattern Name can be any descriptive name. The Grp No. is set to the trunk-group number for the trunk to be used. The FRL field defines the facilty restriction level for this route pattern. The value of 0 is the least restrictive. The Pfx

Mrk field is set to 1. The Prefix Mark sets the requirement for sending a prefix digit 1.

Setting the Pfx Mrk field to a 1 results in a 1 being prefixed to any 10-digit number. An 11-digit number, presumably already with a 1, is left unchanged. Default values for all other fields can be used.

change signaling-group 12 Page 1 of 5 SPE B SIGNALING GROUP

Group Number: 12 Group Type: isdn-pri

Associated Signaling? y Max number of NCA TSC: 0 Primary D-Channel: 01C0724 Max number of CA TSC: 0 Trunk Group for NCA TSC: Trunk Group for Channel Selection: 12 X-Mobility/Wireless Type: NONE Supplementary Service Protocol: a Network Call Transfer? n

change route-pattern 12 Page 1 of 3 Pattern Number: 1 Pattern Name: PSTN

SCCAN? n Secure SIP? n

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4. Configure Avaya SES

This section covers the configuration of Avaya SES. Avaya SES is configured via an Internet browser using the administration web interface. It is assumed that Avaya SES software and the license file have already been installed on the server. During the software installation, the

installation script is run from the Linux shell of the server to specify the IP network properties of the server along with other parameters. For additional information on these installation tasks, refer to [5].

1. Access the Avaya SES administration web interface by entering

http://<ip-addr>/admin as the URL in a Web browser, where <ip-addr> is the IP

address of the Avaya SES server.

Log in with the appropriate credentials and then select the Launch Administration

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4. From the left pane of the administration web interface, expand the Server

Configuration option and select System Properties. The Edit System Properties

page displays the software version in the SES Version field and the network properties entered during the installation process.

On the Edit System Properties page:

Enter the SIP Domain name assigned to Avaya SES. This must match the

Authoritative Domain field configured on Avaya Communication Manager

shown in Section 3.1, Step 5.

Enter the License Host field. This is the host name, the fully qualified domain name, or the IP address of the SIP proxy server that is running the WebLM application and has the associated license file installed.

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5. After setting up the domain on the Edit System Properties page, create a host computer entry for Avaya SES. The following example shows the Edit Host page since the host had already been added to the system.

The Edit Host page shown below is accessible by clicking on the Hosts List link in

the left pane and then clicking on the Edit link under the Commands section of the subsequent page that is displayed.

In the Host IP Address field, enter the IP address of the Avaya SES server. Enter the DB Password that was specified during the system installation. The default values for the other fields may be used.

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6. From the left pane of the administration web interface, expand the Media Servers option and select Add to add the Avaya Media Server to the list of media servers known to Avaya SES. Adding the media server will create the Avaya SES side of the SIP trunk previously created in Avaya Communication Manager.

On the Add Media Server Interface page, enter the following information: Enter a descriptive name in the Media Server Interface Name field (e.g.

S8300).

In the Host field, select the Avaya SES server from the pull-down menu that will serve as the SIP proxy for this media server. Since there is only one Avaya SES server in this configuration, the Host field is set to the host shown in Step 5.

Select TLS (Transport Link Security) for the SIP Trunk Link Type. TLS provides encryption at the transport layer. TLS is the only link protocol that is supported for communication between Avaya SES and Avaya Communication Manager.

Enter the IP address of the Avaya S8300 Media Server in the SIP Trunk IP

Address field. In alternative configurations that use a C-LAN board, the SIP Trunk IP Address would be the IP address of the C-LAN board.

The default values may be retained for all other fields.

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7. A Host Address Map is required on Avaya SES to direct calls outbound from Avaya Communication Manager to the Mediant 1000. The Mediant 1000 does not register as an endpoint with Avaya SES so calls are not automatically routed to the Mediant 1000 based on a registered extension. Instead, an Address Map is used to route calls based on the contents of the SIP INVITE URI matching a specified pattern to determine the proper destination of the call. The URI takes the form of sip:user@domain, where

domain can be a domain name or an IP address. The user portion can be an

alpha-numeric name, telephone number or extension.

In the case of the compliance test, the user portion contained the called party number. All calls bound for the PSTN were routed to the Mediant 1000. Thus, the Host Address Map was configured to match all calls dialed with an 11 digit number beginning with a 1.

To configure a Host Address Map:

Expand the Hosts option in the left pane of the administration web interface and select List. This will display the List Hosts page below.

Click on the Map link associated with the appropriate host to display the List

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8. On the Add Host Address Map page that appears:

Enter a descriptive name in the Name field.

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9. Next, a Host Contact must be entered for the Address Map that was previously defined. The contact defines the destination IP address, port number and transport protocol to use when routing calls that match the Address Map.

To add a Host Contact:

Open the List Host Address Map page as described (but not shown) in Step 7. Click on the Add Another Contact link associated with the address map added

previously to open the Add Host Contact page shown below.

In the Contact field, enter the destination IP address (ip_addr), port number (port) and transport protocol (protocol) in the following format.

sip:$(user)@ip_addr:port;transport=protocol The user part in the original request URI is inserted in place of the “$(user)” string before the message is sent to the destination.

For the compliance test, the Mediant 1000 had IP address of 192.168.1.60. Thus, the following contact value was used:

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10. After configuring the Host Address Map and Contact the List Host Address Map page will appear as shown below.

11. A Media Server Address Map is required on Avaya SES to direct calls inbound to Avaya Communication Manager from the Mediant 1000 in the same way that

outbound calls from Avaya Communication Manager required a Host Address Map. In the case of the compliance test, calls from the PSTN beginning with 1732555 were routed through the Avaya DEFINITY Server R and then from the Mediant 1000 to Avaya Communication Manager at Site 1. The leading 1 was deleted by the Avaya DEFINITY Server R at Site 2 before sending the call to the Mediant 1000. Thus, the Media Server Address Map was configured to match all calls dialed with a 10 digit number beginning with 732555.

To configure a Media Server Address Map:

Expand the Media Servers option in the left pane of the administration web interface and select List. This will display the List Media Servers page below. Click on the Map link associated with the appropriate host to display the List

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12. On the Add Media Server Address Map page that appears: Enter a descriptive name in the Name field.

In the Pattern field, enter an expression to define the matching criteria for calls to be routed from the Mediant 1000 to Avaya Communication Manager. The example below shows the expression used in the compliance test. This expression will match an URI that begins with sip:732555 followed by any digit between 0-9 for the next 4 digits. Appendix A contains additional information on the syntax used for address map patterns.

Click the Add button.

13. After configuring the Media Server Address Map, the List Media Server Address

Map page appears as shown below. The first Media Server Contact is created

automatically and directs the calls to the IP address of the Avaya Media Server

(50.1.1.10) using port 5061 and TLS as the transport protocol. The user portion in the original request URI is substituted for “$(user)”. For the compliance test, the Contact field for the Media Server Address Map is displayed as:

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14. Lastly, the IP address of the Mediant 1000 must be configured as a trusted host on Avaya SES. As a trusted host, Avaya SES will not issue SIP authentication challenges for incoming requests from the designated IP address.

To configure a trusted host:

Connect to Avaya SES and log in using proper credentials.

Enter the following trustedhost command at the Linux shell prompt.

trustedhost –a 192.168.1.60 –n 50.1.1.50 –c Mediant1000

Use the following trustedhost command to verify the entry is correct.

trustedhost –L

Important Note: Complete the trusted host configuration by returning to the main Avaya SES administration web interface and clicking the Update link as shown in Step 3. If the Update link is not visible, refresh the page by selecting the Top link from the left menu. This step is required even though the trusted host was configured via the Linux shell.

The screen below illustrates the results of the trustedhost commands.

admin@denali> trustedhost –a 192.168.1.60 –n 50.1.1.50 –c Mediant1000 192.168.1.60 is added to trusted host file list.

admin@denali> trustedhost -L Third party trusted hosts.

Trusted Host | CCS Host Name | Comment

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5. Configure the Mediant 1000

This section describes the procedures for configuring the Mediant 1000. These procedures assume the Mediant 1000 has been installed using the procedures documented in [7] and has been assigned an IP address.

1. The configuration of the Mediant 1000 is done via a Web browser. To access the device, enter the IP address of the Mediant 1000 in the Address field of the browser.

2. The following pop-up window will appear. Login with the proper credentials.

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4. The network settings that were configured during installation can be viewed by selecting Advanced Configuration in the left pane then navigating to Network

Settings IP Settings in the right pane. If necessary, changes can be made to the

settings on this page followed by clicking Submit. For the compliance test, the IP

Address, Subnet Mask and Default Gateway Address were set to values consistent

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5. Protocol Definition

To access these parameters, select Protocol Management in the left pane and navigate to Protocol Definition in the right pane. The pull-down choices for Protocol

Definition are shown below.

6. Proxy and Registration

From the menu shown in Step 5, navigate to Protocol Definition Proxy & Registration. Configure the parameters as described below.

For the Enable Proxy field, select Use Proxy from the pull-down menu. In the Proxy IP Address field, enter the IP address of the Avaya SES server. For the Always Use Proxy field, select Enabled.

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7. General Protocol Parameters

From the menu shown in Step 5, navigate to Protocol Definition General Parameters. Configure the parameters as described below.

For the Enable Early Media field, select Enabled. If enabled, the Mediant 1000 sends Session Description Protocol (SDP) information in the 18x SIP responses allowing the media stream to be set-up prior to answering the call. If fax support is needed, select T.38 Relay as the Fax Signaling Method. Select No for the Use “user=phone” in SIP URL field.

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8. Codecs

From the menu shown in Step 5, navigate to Protocol Definition Coders. In the

screen below, select the list of preferred codecs to be used by the Mediant 1000 with the most preferred codec at the top and working downward to the least preferred. This list must have an overlap with the list provided on Avaya Communication Manager in Section 3.1, Step 6. The codec is selected from the pull-down menu under the Coder

Name field.

The codec list used for the compliance test is shown in the example below.

G.711U-law was selected as the most preferred followed by G.729. For the G.729 codec, the

Silence Suppression field was set to Enable. With silence suppression enabled, the

Mediant 1000 uses the equivalent of a G.729AB codec. Default values were retained for all other fields.

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9. DTMF and Dialing

From the menu shown in Step 5, navigate to Protocol Definition DTMF & Dialing. Configure the parameters as described below.

In the Max Digits in Phone Num field, enter the maximum number of digits that can be dialed.

For the Declare RFC 2833 in SDP field, select Yes. For the 1st

Tx DTMF Option field, select RFC 2833. This selects RFC 2833 as

the preferred DTMF transmission method.

Select 127 as the RFC 2833 Payload Type to match the value used by the Avaya SIP Telephones. Media may not be redirected (shuffled) in all scenarios from Avaya Communication Manager to the endpoints if this value is not the same as the SIP Telephones.

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10. Advanced Parameters

To access these parameters, select Protocol Management in the left pane and navigate to Advanced Parameters in the right pane. The pull-down choices for Advanced

Parameters are shown below.

11. General Advanced Parameters

From the menu shown in Step 10, navigate to Advanced Parameters General Parameters. Configure the parameters as described below.

In the Max Number of Active Calls field, verify the value is set to the maximum value of 150.

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12. Trunk Group

Select Protocol Management in the left pane and navigate to Trunk Group in the right pane.

The Trunk Group Table maps a particular trunk channel to a trunk group. In the From

Trunk and To Trunk columns, enter the starting and ending trunks to be assigned. In

the Channel(s) column, enter the range of channels on those trunks to be assigned. A phone number may be entered in the Phone Number column or it may be left blank. If a number is entered, this number will be used as the originating calling party if no calling party information is received from the originating PSTN trunk. Each channel is assigned a unique number starting with the value in the Phone Number column and incrementing for each subsequent channel. If the Phone Number column is left blank, the Mediant 1000 will use a default value for the originating calling party if no calling party information is received from the originating PSTN trunk. In the Trunk Group

ID column, enter the trunk group that will contain these channels. The default value

may be used for the Profile ID column.

In the example below, the table entry assigns channels 1 – 24 of trunk 1 to Trunk Group 1. A range of numbers arbitrary chosen to start at 44444 will be used for the originating calling party number if no calling party information is received from the originating PSTN trunk.

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13. Trunk Group Settings

Select Protocol Management in the left pane and navigate to Trunk Group Settings in the right pane.

Configure the parameters as described below.

For Trunk Group ID 1 which contains the T1 channels, select the Channel

Select Mode as Cyclic Ascending. The channels in this trunk group are treated

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14. Routing Tables

To access these parameters, select Protocol Management in the left pane and navigate to Routing Tables in the right pane. The pull-down choices for Routing Tables are shown below.

15. Telephony to IP Routing

From the menu shown in Step 14, navigate to Routing Tables Tel to IP Routing.

This table defines the mapping of PSTN trunk calls to an IP address that will process the IP leg of the call. The Dest. Phone Prefix and Source Phone Prefix columns define which calls are mapped to the IP address in the Dest. IP Address column. In the example below, the table entry maps calls from any destination prefix, or any source prefix to IP address 50.1.1.0 which is the address of the Avaya SES server. The default values may be used for the Profile ID column.

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16. IP to Hunt Group Tables

From the menu shown in Step 14, navigate to Routing Tables IP to Hunt Group Routing.

This table defines the mapping of IP calls to the trunk group created in Step 12. The

Dest. Phone Prefix, Source Phone Prefix and Source IP Address columns define

which calls are mapped to the trunk group in the Trunk Group ID column. In the example below, the table entry maps calls from any destination prefix, or any source prefix or any source IP address to trunk group 1.

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17. Media Settings

To access these parameters, select Advanced Configuration in the left pane and navigate to Media Settings in the right pane. The pull-down choices for Media

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18. Fax/Modem/CID Settings

From the menu shown in Step 17, navigate to Media Settings Fax/Modem/CID Settings. Configure the parameters as described below.

Select T.38 Relay for the Fax Transport Mode to support faxing across the data WAN.

Select Enable Bypass for the V.22 – V.34 Modem Transport Types to support modem calls over SIP planned in future Avaya Communication Manager releases.

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19. Trunk Settings

Select Advanced Configuration in the left pane and navigate to Trunk Settings in the right pane. These parameters will vary based on the trunk settings provided by the far-end. For the compliance test, these parameters must be compatible with the settings used on Avaya Communication Manager in Section 3.2, Step 3. The parameters were configured as described below.

The Protocol Type was set to T1 5ESS 10 ISDN. Avaya Communication Manager was also configured to use the 5ESS version of ISDN by the proper selection of the Country Protocol and Protocol Version fields on the DS1

Circuit Pack form.

The Line Code was set to B8ZS. This must match the corresponding value on Avaya Communication Manager.

The Framing Method was set to T1 Framing ESF CRC6. This must match the corresponding value on Avaya Communication Manager.

The ISDN Termination Side was set to User side because the Avaya

Communication Manager side of the link was set to network. One side must be network and the other side must be user.

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20. For proper interoperability, two additional parameters must be changed in the ini file. From a web browser, enter <ip_address>/AdminPage in the Address field where <ip_address> is the IP address of the Mediant 1000. The main page will appear as shown below.

21. Navigate to ini Parameters in the left pane. In the Parameter Name field, select

ModemBypassPayloadType from the pull-down menu. Enter 0 in the Enter Value

field. Click on Apply New Value.

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22. Navigate to ini Parameters in the left pane. In the Parameter Name field, select

ISUSETOHEADERASCALLEDNUMBER from the pull-down menu. Enter 1 in the

Enter Value field. Click on Apply New Value.

For telephony to IP calls, the Mediant 1000 can be configured to change the content of the Contact header in the SIP messages it generates for these calls. This is done by changing the value of the ISUSETOHEADERASCALLEDNUMBER parameter. By default, this parameter is set to 0 and the originating PSTN calling number is used in the Contact header. If this parameter is set to 1, then the gateway extension is used in the Contact header. For interoperability, this parameter must be set to 1.

23. To save the configuration, return to the administrative web page and select

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6. Interoperability Compliance Testing

This section describes the compliance testing used to verify the interoperability between the AudioCodes Mediant 1000, Avaya SIP Enablement Services (SES) and Avaya Communication Manager. This section covers the general test approach and the test results.

6.1. General Test Approach

The general test approach was to make calls to/from the PSTN through the Mediant 1000 at Site 2 using various codec settings, telephone types and exercising common PBX features.

6.2. Test Results

The AudioCodes Mediant 1000 successfully passed compliance testing. The following features and functionality were verified for calls passing through the Mediant 1000. PBX features were initiated from endpoints at Site 1 and performed on calls passing through the Mediant 1000.

Calls to/from the PSTN

G.711mu and G.729AB codec support Proper recognition of DTMF transmissions

Support for Hold, Transfer, Conferencing, Call Forwarding and Call Waiting Proper operation of voicemail with message waiting indicators (MWI).

Extended telephony features using Avaya Communication Manager Feature Name

Extensions such as Call Park, Call Pickup and Last Number Dialed. For more information on FNEs, please refer to [6].

T.38 fax support

Proper system recovery after a Mediant 1000 restart

The following observations were made during the compliance test:

To support T.38 faxing, a post-GA release of Avaya Communication Manager was required.

7. Verification Steps

The following steps may be used to verify the configuration:

• From the Avaya Communication Manager SAT, use the status signaling-group command to verify that the SIP signaling group is in-service.

• From the Avaya Communication Manager SAT, use the status trunk-group command to verify that the SIP trunk group is in-service.

• Verify that calls can be placed to/from the PSTN through the Mediant 1000.

8. Support

For technical support on the Mediant 1000, contact AudioCodes via the support link at

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9. Conclusion

These Application Notes describe the procedures required to configure the AudioCodes Mediant 1000 VoIP Media Gateway to interoperate with Avaya SIP Enablement Services and Avaya Communication Manager. The AudioCodes Mediant 1000 successfully passed compliance testing with the observations documented in Section 6.2.

10. Additional References

[1] Feature Description and Implementation For Avaya Communication Manager, Doc # 555-245-205, Issue 4.0, February 2006

[2] Administrator Guide for Avaya Communication Manager, Doc # 03-300509, Issue 2.1, May 2006

[3] Avaya Communication Manager Advanced Administration Quick Reference, Doc # 03-300364, Issue 2, June 2005 Release 3.0

[4] Avaya IA 770 INTUITY AUDIX Messaging Application, Doc # 11-300532, May 2005 [5] Installing and Administering SIP Enablement Services R3.1, Doc# 03-600768, Issue 1.5,

February 2006

[6] Avaya Extension to Cellular and Off-PBX Station (OPS) Installation and Administration Guide

Release 3.0, version 6.0, Doc # 210-100-500, Issue 9, June 2005

[7] LTRT-68804 Mediant 2000 and Mediant 1000 TP-1610 and TP-260 Boards User Manual 4.8 [8] LTRT-65607 MediaPack & Mediant 1000 SIP Analog Gateways Release Notes, Version 5.0. Product documentation for Avaya products may be found at http://support.avaya.com.

Product documentation for AudioCodes Mediant 1000 VoIP Media Gateway products may be found

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APPENDIX A: Specifying Pattern Strings in Address Maps

The syntax for the pattern matching used within Avaya SES is a Linux regular expression used to match against the URI string found in the SIP INVITE message.

Regular expressions are a way to describe text through pattern matching. The regular expression is a string containing a combination of normal text characters, which match themselves, and special

metacharacters, which may represent items like quantity, location or types of character(s).

In the pattern matching string used in Avaya SES:

Normal text characters and numbers match themselves. Common metacharacters used are:

o A period . matches any character once (and only once). o A asterisk * matches zero or more of the preceding characters.

o Square brackets enclose a list of any character to the matched. Ranges are designated by using a hyphen. Thus, the expression [12345] or [1-5] both describe a pattern that will match any single digit between 1 and 5.

o Curley brackets containing an integer ‘n’ indicate that the preceding character must be matched exactly ‘n’ time. Thus, 5{3} matches ‘555’ and [0-9]{10} indicates any 10 digit number.

o The circumflex character ^ as the first character in the pattern indicates that the string must begin with the character following the circumflex.

Putting these constructs together as used in this document, the pattern to match the SIP INVITE string for any valid 1+ 10 digit number in the North American dial plan would be:

^sip:1[0-9]{10}

This reads as: “Strings that begin with exactly sip:1 and having any 10 digits following will match. A typical INVITE request below uses the shaded portion to illustrate the matching pattern.

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presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes.

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