ShoreTel / AudioCodes

960 Stewart Drive Sunnyvale, CA 94085 USA Phone +1.408.331.3300 +1.877.80SHORE Fax +1.408.331.3333 www.ShoreTel.com

Application Note

ShoreTel / AudioCodes

Enterprise Session Border Controllers for SIP Trunking

Version 1 Issue 1

I n n o v a t i o n

N e t w o r k

A p p N o t e

IN- 10448 Date: August, 2012

Product: ShoreTel | AudioCodes System version:

ShoreTel 12.2


Table of Contents

APPLICATION NOTE .......................................................................................................................... 1

SHORETEL / AUDIOCODES .............................................................................................................. 1

ENTERPRISE SESSION BORDER CONTROLLERS FOR SIP TRUNKING .............................. 1

1 OVERVIEW .................................................................................................................................. 3

1.1 DOCUMENT FEEDBACK ......................................................................................................... 4 1.2 DOCUMENT CHANGE HISTORY ............................................................................................. 4 1.3 SIP TRUNKING ARCHITECTURE OVERVIEW.......................................................................... 4

2 INTEROPERABILITY REQUIREMENTS, CERTIFICATION AND LIMITATIONS ............ 5

2.1 VERSION SUPPORT ............................................................................................................... 5 2.2 LIMITATIONS .......................................................................................................................... 5 2.3 LIST OF COMPLIANT HARDWARE............................................................................................ 5

2.3.1 ShoreTel System ............................................................................................................ 5 2.3.2 AudioCodes .................................................................................................................... 5 2.3.3 Voice switches ................................................................................................................ 5 2.3.4 Phones............................................................................................................................. 6

2.4 CERTIFICATION RESULTS SUMMARY .................................................................................... 6 TABLE 1: INITIALIZATION AND BASIC CALLS ........................................................................................ 7 TABLE 2: MEDIA AND DUAL-TONE MULTI-FREQUENCY (DTMF) SUPPORT ...................................... 7 TABLE 3: PERFORMANCE & QUALITY OF SERVICE ............................................................................. 8 TABLE 4: ENHANCED SERVICES AND FEATURES ................................................................................ 9 TABLE 5: SECURITY ........................................................................................................................... 10

3 SHORETEL ................................................................................................................................. 11

3.1 SHORETEL UNSUPPORTED FEATURES............................................................................... 11 3.2 SHORETEL CONFIGURATION............................................................................................... 12

3.2.1 Call Control Settings .................................................................................................... 12 3.2.2 Sites Settings ................................................................................................................ 14 3.2.3 Switch Settings - Allocating Ports .............................................................................. 16 3.2.4 System Settings – Trunk Groups ............................................................................... 18 3.2.5 System Settings – Individual Trunks ......................................................................... 21

4 AUDIOCODES ........................................................................................................................... 23

4.1 AUDIOCODES ENTERPRISE SBC (E-SBC) PRODUCT INFORMATION................................ 23 4.2 AUDIOCODES PRODUCT CONFIGURATION ......................................................................... 23 4.3 CONFIGURATION OF THE AUDIOCODES ESBC PRODUCTS ................................................ 24

DOCUMENT AND SOFTWARE COPYRIGHTS ............................................................................ 44

5 TRADEMARKS .......................................................................................................................... 44

6 DISCLAIMER .............................................................................................................................. 44

7 COMPANY INFORMATION ..................................................................................................... 44


ShoreTel tests and validates the interoperability of the Member's solution with ShoreTel's published software interfaces. ShoreTel does not test, nor vouch for the Member's development and/or quality assurance process, nor the overall feature functionality of the Member's solution(s). ShoreTel does not test the Member's solution under load or assess the scalability of the Member's solution. It is the responsibility of the Member to ensure their solution is current with ShoreTel's published interfaces. The ShoreTel Technical Support organization will provide Customers with support of ShoreTel's published software interfaces. This does not imply any support for the Member's solution directly. Customers or reseller partners will need to work directly with the Member to obtain support for their solution.

1 Overview This document provides details for connecting the ShoreTel UC system though the AudioCodes Mediant gateway products to ITSP’s for SIP Trunking (in this case with Optus) to enable audio communications. The document specifically focuses on the configuration procedures needed to set up these systems to interoperate. SIP Trunking allows the use of Session Initiation Protocol (SIP) communications from an Internet Telephony Service Provider (ITSP) instead of the typical analog, Basic Rate Interface (BRI), T1 or E1 trunk connections. Having the pure IP trunk to the ITSP allows for more control and options over the communication link. This application note provides the details on connecting the ShoreTel® IP phone system through an AudioCodes Mediant 1000 eSBC which acts as a secure gateway and in this case, for testing purposes, is connected to both to Optus Australia for SIP Trunking. ShoreTel and AudioCodes have teamed up to build a solid security focused solution, ShoreTel being the IP PBX which sits on the LAN and connects to the AudioCodes SBC. Providing a solution to allow customers the ability to connect to SIP Trunks. The AudioCodes SBC is connected to not only the LAN but also the WAN, providing the typical firewall security abilities but also intelligent SIP routing and such SIP features as:

Registration

Digest Authentication

Dial Plan Modification

Back to Back User Agent (Terminates SIP messaging on both LAN and WAN side for SIP Protocol Normalization)

Transfer conversion of SIP REFER to SIP reINVITE messaging

Quick configuration templates for each of the certified ITSPs

Note: This application note is specific to ShoreTel and AudioCodes. While AudioCodes supports many Service Providers via the eSBC, ShoreTel has currently only Validated the configuration with Optus and Telstra. If the ShoreTel customer wishes to connect the solution to an ITSP that has not been Validated by ShoreTel, the end customer will need to get support from AudioCodes. ShoreTel will support the interface between ShoreTel and AudioCodes, but non-Validated ITSPs will require direct support form AudioCodes.


1.1 Document Feedback ShoreTel IP PBX administrators who would like to provide feedback on the contents of this document should send it to [email protected].

1.2 Document Change History Version 1 Issue 1 07/24/2012; Initial Draft

1.3 SIP Trunking Architecture Overview

There are a number of different network deployments of SIP Trunking from a Service Provider to an Enterprise. Some Service Providers provide SIP Trunking directly over the Internet and others over provide managed link into their network. Here are two typical deployments examples:

AudioCodes LAN IP Address

10.195.0.47

Optus

Evolve SIP Trunks

Optus WAN IP Address

123.102.83.115

Optus SIP Trunks

123.102.2.131

ShoreTel SIP Trunks 10.195.0.45

mailto:[email protected]


2 Interoperability Requirements, Certification and Limitations

2.1 Version Support Products are certified via the ShoreTel Innovation Network Certification Process for the ShoreTel system. The table below contains the matrix of AudioCodes versions firmware releases certified on the identified ShoreTel software releases.

AudioCodes eSBC version

6.40A.016.003

ShoreTel 12.2

2.2 Limitations While AudioCodes supports many Service Providers via the eSBC, ShoreTel has currently only Validated the configuration with Optus and Telstra. If the ShoreTel customer wishes to connect the solution to an ITSP that has not been Validated by ShoreTel, the end customer will need to get support from AudioCodes. ShoreTel will support the interface between ShoreTel and AudioCodes, but non-Validated ITSPs will require direct support form AudioCodes.

2.3 List of compliant hardware The following hardware while not necessarily used during the certification itself is guaranteed to comply with the testing results of this application note.

2.3.1 ShoreTel System

The ShoreTel system used during the certification process is a ShoreWare Director Enterprise Edition Server consisting of ShoreTel 12.2 version. .

2.3.2 AudioCodes

The AudioCodes SBC system used during certification is a Mediant 1000 eSBC consisting of Firmware version 6.40A.016.003 AudioCodes’ Mediant 800, Mediant 1000™ and Mediant 3000 Enterprise Session Border Controllers (E-SBC) enable connectivity and security between enterprises and Service Providers’ VoIP networks. AudioCodes’ Mediant E-SBCs are designed as a secured VoIP and data platform. Enhanced Media Gateway security features include encryption schemes, such as SRTP for media, TLS for SIP control, IPSec for management and Denial of service protection. A fully featured Enterprise-class Session Border Controller provides a secured voice network deployment, based on the embedded Back-to-Back User Agent (B2BUA).

2.3.3 Voice switches

ShoreGear 30 ShoreGear 50


ShoreGear 50v ShoreGear 90 ShoreGear 90v ShoreGear 90BRI ShoreGear 90BRIv ShoreGear 120 ShoreGear 220T1 ShoreGear 220T1a ShoreGear 220E1

2.3.4 Phones

Regardless the technology of phones connected to the ShoreTel system, all officially supported phones may place calls over a Optus SIP trunk & will demonstrate the same behavior (features & limitations) as captured in this application note.

2.3.4.1 ShoreTel IP phones

ShorePhone IP 110 ShorePhone IP115 ShorePhone IP212k ShorePhone IP230 ShorePhone IP230g ShorePhone IP265 ShorePhone IP560 ShorePhone IP560g ShorePhone IP565 ShorePhone IP655 ShorePhone IP8000 ShorePhone BB24

2.3.4.2 SIP phones

All officially ShoreTel Innovation Network Validated SIP phones are also compliant with the test results documented in this application note.

2.3.4.3 Analog Phones

Any analog phones connected to the following ShoreGear voice switches will be compliant with the test results documented in this application note. ShoreGear 30 ShoreGear 50 ShoreGear 50v ShoreGear 90 ShoreGear 90v ShoreGear 90BRI ShoreGear 90BRIv ShoreGear 120 ShoreGear 220T1a

2.4 Certification Results Summary Test Plan for AudioCodes leveraging the Optus SIP Trunking Service


Table 1: Initialization and Basic Calls

ID Name Description Notes

1.1 Setup and initialization

Verify successful setup and initialization of the system under test (SUT)

Pass

1.2 Outbound Call (Domestic)

Verify calls outbound placed through the SUT reach the external destination.

Pass

1.3 Inbound Call (Domestic)

Verify calls received by the SUT are routed to the default trunk group destination.

Pass

1.4 Device Restart –Power Loss

Verify that the SUT recovers after power loss to the SUT

Pass

1.5 Device Restart – Network Loss

Verify the SUT recovers after loss of network link to the SUT.

Pass

1.6 All Trunks Busy – Inbound Callers

Verify an inbound caller hears busy tone when all channels/trunks are in use

Pass

1.7 All Trunks Busy – Outbound Callers

Verify an outbound caller hears busy tone when all channels/trunks are in use

Pass

1.8 Incomplete Inbound Calls

Verify proper call progress tones are provided and proper call teardown for incomplete inbound calls.

Pass

Table 2: Media and Dual-Tone Multi-Frequency (DTMF) Support


2.1 Media Support – ShoreTel Phone to SUT

Verify call connection and audio path from a ShoreTel phone to an external destination through the service provider using all supported codecs with both sides set to a common codec.

Pass Note 1

2.2 Media Support – SIP Reference to SUT

Verify call connection and audio path from SIP Reference phones to an external destination through the service provider using all supported codes with both sides set to a common codec.

Pass

2.3 Codec Negotiation

Verify codec negotiation between the SUT and the calling device with each side configured for a different codec.

Pass

2.4 DTMF Transmission – Out of Band / In Band

Verify transmission of in-band and out-of-band digits per RFC 2833 for various devices connected to the SUT.

Pass

2.5 Auto Attendant Menu

Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the desired extension.

Pass

2.6 Auto Attendant Menu “Dial by Name”

Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the desired extension using the “Dial by Name” feature.

Pass

2.7 Auto Attendant Menu Checking Voice Mail Mailbox

Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the Voice Mail Login Extension.

Pass


Table 3: Performance & Quality of Service


3.1 Voice Quality Service Levels

Verify the SUT can provide a voice quality service-level agreement (SLA) across the WAN from the customer premises to the SUT SIP gateway.

Pass

3.2 Capacity Test Verify the service provider interface can sustain services through period of heavy outbound and inbound load.

Pass

3.3 Post Dial Delay Verify that post dial delay is within acceptable limits.

Pass

3.4 Billing Accuracy Verify that all test calls made are accurately reflected in the SUT’s Call Detail Record (CDR) and billing reports.

Pass


Table 4: Enhanced Services and Features


4.1 Caller ID Name and Number - Inbound

Verify that Caller ID name and number is received from SIP endpoint device

Pass

4.2 Caller ID Name and Number - Outbound

Verify that Caller ID name and number is sent from SIP endpoint device

Pass

4.3 Hold from SUT to SIP Reference

Verify successful hold and resume of connected call

Pass

4.4 Call Forward - SUT

Verify outbound calls that are being forwarded by the SUT are redirected and connected to the appropriate destination.

Pass

4.5 Call Transfer – Blind

Verify a call connected from the SUT to the ShoreTel phone can be transferred to an alternate destination.

Pass

4.6 Call Transfer – Consultative

Verify a call connected from the SUT to the ShoreTel phone can be transferred to an alternate destination.

Pass

4.7 Conference – Ad Hoc

Verify successful ad hoc conference of three parties

Pass

4.8 Inbound Direct Inward Dialing/Dialed Number Identification Service (DID/DNIS)

Verify the SUT provides inbound “dialed number information” and is correctly routed to the configured destination.

Pass

4.9 Outbound 911 Verify that outbound calls to 911 are routed to the correct Public Safety Answering Point (PSAP) for the calling location and that caller ID information is delivered.

Not Tested Note 2

4.10 Operator Assisted

Verify that calls are routed to an operator for calling assistance.

Pass

4.11 Inbound / Outbound call with Blocked Caller ID

Verify that calls with Blocked Caller ID route properly and the answering phone does not display any Caller ID information.

Pass

4.12 Inbound call to a Hunt Group

Verify that calls route to the proper hunt group and are answered by an available hunt group member with audio in both directions using G.729 and G.711 codecs.

Pass

4.13 Inbound Call to a Workgroup

Verify that calls route to the proper workgroup and are answered successfully by an available workgroup agent with audio in both directions using G.729 and G.711 codecs.

Pass

4.14 Inbound Call to DNIS/DID and Leave a Voice Mail Message

Verify that inbound calls to a user, via DID/DNIS, routes to the proper user mailbox and a message can be left with proper audio.

Pass

4.15 Call Forward – “FindMe”

Verify that inbound calls are forwarded to a user’s “FindMe” destination.

Pass

4.16 Call Forward Always

Verify that inbound calls are immediately automatically forwarded to a user’s external destination.

Pass

4.17 Inbound / Outbound Fax Calls

Verify that inbound/outbound fax calls complete successfully.

Pass



4.18 ShoreTel Converged Conferencing Server

Verify that inbound calls are properly forwarded to the ShoreTel Converged Conferencing Server, that it properly accepts the access code, and you’re able to participate in the conference bridge.

Pass

4.19 Inbound Call to Bridged Call Appearance (BCA) Extension

Verify that inbound calls properly presented to all of the phones that have BCA configured and that the call can be answered, placed on-hold and then transferred.

Pass

4.20 Inbound Call to a Group Pickup Extension

Verify that inbound calls are properly presented to all of the phones that have Group Pickup configured and that the call can be answered, placed on-hold and then transferred.

Pass

Table 5: Security


5.1 Digest Authentication

Verify the SUT supports the use of digest authentication for service access for inbound and outbound calls.

N/A

Note 1: Analog phones connected to ShoreTel ShoreGear switches do not currently support G.711a-law (PCMA) codecs. Note 2: Emergency Services calls were conducted in a controlled environment and not actually sent via Optus SIP trunks.


3 ShoreTel The configuration information below shows examples for configuring the ShoreTel, AudioCodes and Optus SIP Trunking network. Even though configuration requirements can vary from setup to setup, the information provided in these steps, along with the ShoreTel Planning and Installation Guide (or the Administration Guide) and documentation provided by AudioCodes and Optus should prove to be sufficient. However every design can vary and some may require more planning than others. Note: while AudioCodes supports many Service Providers via the eSBC, ShoreTel has currently only Validated the configuration with Optus and Telstra. If the ShoreTel customer wishes to connect the solution to an ITSP that has not been Validated by ShoreTel, the end customer will need to get support from AudioCodes. ShoreTel will support the interface between ShoreTel and AudioCodes, but non-Validated ITSPs will require direct support form AudioCodes.

3.1 ShoreTel Unsupported Features Please consult the ShoreTel Administration Guide, Section 18 – Session Initiation Protocol, and more specifically Section 18.1.2.5 and 18.1.2.6 for specific details regarding unsupported features. Here is an excerpt of the Section 18.1.2.6: General Feature Limitations

ShoreWare supports Music On Hold (MOH) over SIP trunks. The capacity limits of MOH switches is the same as other trunks; a switch can provide up to 15 streams. However, these streams can be to other switches or to SIP devices.

If the ShoreTel server has a conference bridge 4.2 installed, you should not enable SIP. The conference bridge is not compatible with a ShoreTel system that has SIP enabled due to the dynamic RTP port required for SIP.

3-way conference on a SIP trunk call uses Make Me conference ports. A minimum of 3 Make Me ports must be configured to support 3-way conferencing. Make Me conferencing for 4 to 6 parties is not supported.

A SIP trunk can be a member of a 3-party conference but cannot initiate a 3-way conference (unless the SIP device merges the media streams itself).

ShoreTel SIP supports basic transfers (i.e. blind transfers) and attended transfers (i.e. consultative transfers).

Silent Monitoring is not supported on a SIP trunk call.

Barge-In is not supported on a SIP trunk call.

Call recording is not supported on a SIP trunk call. Call recording requires presence of a physical trunk in the call.

Call redirection by SIP devices is not supported.

Park/Unpark is not supported on a SIP trunk call. This is planned for a future release.

Silence detection on trunk-to-trunk transfers is not supported since it requires a physical trunk.

Fax (and modem) redirection is not supported with SIP trunks as only physical trunks can detect fax tones.


3.2 ShoreTel Configuration

This section describes the ShoreTel system configuration to support SIP Trunking. The section is divided into general system settings and trunk configurations (both group and individual) needed to support SIP Trunking. Note: ShoreTel just points its Individual SIP Trunks to the AudioCodes eSBC. The first settings to address within the ShoreTel system are the general system settings. These configurations include the Call Control, the Site and the Switch settings. If these items have already been configured on the system, skip this section and go on to the “ShoreTel System Settings – Trunk Groups” section below.

3.2.1 Call Control Settings The first settings to configure within ShoreWare Director are the Call Control Options. To configure these settings for the ShoreTel system, log into ShoreWare Director and select “Administration” then “Call Control” followed by “Options”.

1. Open the 'ShoreWare Director Admin' page click on (Administration > Call Control > Options):


2. The “Call Control Options” screen will then appear

3. Within the “Call Control Options” screen, confirm that the appropriate settings are made for the “DTMF Payload Type”, “Realm”, “Enable SIP Session Timer” and “Always Use Port 5004 for RTP” fields.

4. The “DTMF Payload Type (96-127)” parameter defaults to a value of 102, you will need to ensure that you change this parameter to a value of 101 to interoperate with Optus. Once you modify this parameter you will need to reboot all of the ShoreTel IP Phones, not rebooting the ShoreTel IP phones will cause the default value (102) to be utilized.

5. The “Realm” parameter is used in authenticating all SIP devices. It is typically a description of the computer or system being accessed. Changing this value will require a reboot of all ShoreGear switches serving SIP extensions. It is not necessary to modify this parameter to get the ShoreTel IP PBX system functional with Optus. Verify that the “Enable SIP Session Timer” box is checked (enabled).

6. Next the Session Interval Timer needs to be set. The recommended setting for “Session Interval” is 3600 seconds. The last item to select is the appropriate refresher (from the pull down menu) for the SIP Session Timer. The “Refresher” field will be set either to “Caller (UAC)” [User Agent Client] or to “Callee (UAS)” [User Agent Server]. If the “Refresher” field is set to “Caller (UAC)”, the Caller’s device will be in control of the session timer refresh. If “Refresher” is set to “Callee (UAS)”, the device of the person called will control the session timer refresh.

7. The last item is to verify that the “Always Use Port 5004 for RTP” is not enabled, if this is a new installation the option will be grayed out, do not modify this parameter if it is disabled.

Note: Disabling (un-checking) the parameter “Always Use Port 5004 for RTP” is required for implementing SIP on the ShoreTel system. For SIP configurations, Dynamic User Datagram Protocol (UDP) must be used for RTP Traffic. If the box is unchecked, MGCP will no longer use UDP port 5004; MGCP and SIP traffic will use dynamic UDP ports. Once this parameter is unchecked, make sure that “everything” (IP Phones, ShoreGear Switches, ShoreWare Director, Distributed Voice Services / Remote Servers, Conference Bridges and Contact Centers) are “fully” rebooted – this is a “one time only” item. By not performing a full system reboot, one way audio will probably occur during initial testing.


3.2.2 Sites Settings

The next settings to address are the administration of sites. These settings are modified under the ShoreWare Director by selecting “Administration”, then “Sites”.

1. Open the 'ShoreWare Director Admin' page click on (Administration > Sites):

2. This selection brings up the “Sites” screen. Within the “Sites” screen, select the name of the site to configure. The “Edit Site” screen will then appear.

Note: Bandwidth of 2046 is just an example. Please refer to the ShoreTel Planning and Installation

Guide for additional information on setting Admission Control Bandwidth.

Sites Edit screen – Admission Control Bandwidth The Admission Control Bandwidth defines the bandwidth available to and from the site. This is important as SIP trunk calls may be counted against the site bandwidth. Bandwidth needs to be set appropriately based on site setup and configuration with Optus. Please refer to the ShoreTel Planning and Installation Guide for additional information. Sites Edit screen – Intra / Inter-Site Calls By default ShoreTel has 11 built-in codecs, these codecs can be grouped as “Codec Lists” and defined in the sites page for “Inter-site” and “Intra-site” calls. By default "Very Low Bandwidth Codecs" contains two codecs, G.729 and G.711u, with G.729 being the primary codec of choice. Make a copy of the “Very Low Bandwidth” codec list, define a name and add the G.711a (PCMA) codec, move it above the PCMU entry, then save the change. It should look as follows:


Then configure the "Inter-Site Calls" option for the “Codec List” you just created and save the

change.

By default the “Very High Bandwidth Codecs” list contains 8 different codecs. Make a copy of the

“Very High Bandwidth Codecs” codec list, define a name and add the G.711a (PCMA) codec, once

again, move it above the PCMU entry, then save the change. It should look as follows:

Then configure the "Intra-Site Calls" option for the “Codec List” you just created and save the

change.

Note: If you wish to use the G.729 codec as your primary choice, select the G729/8000 codec and

move it up above the PCMA/8000 entry.


3.2.3 Switch Settings - Allocating Ports

The final general settings to input are the ShoreGear switch settings. These changes are modified by selecting “Administration”, then “Switches” in ShoreWare Director.

1. Open the 'ShoreWare Director Admin' page click on (Administration > Platform Hardware > Voice Switches > Appliances > Primary): This action brings up the

“Switches” screen

2. From the “Switches” screen simply select the name of the switch to configure.

3. The “Edit ShoreGear …Switch” screen will be displayed. Within the “Edit ShoreGear …Switch” screen, select the desired number of SIP Trunks from the ports available. Each port designated as a SIP Trunk enables the support for 5 individual trunks.


4. Each port designated as a SIP Trunk enables the support for 5 individual trunks.


3.2.4 System Settings – Trunk Groups

If the SIP Trunk Groups have already been configured on the system, skip down to the “ShoreTel System Settings - Individual Trunks” section. The settings for Trunk Groups are changed by selecting “Administration”, then “Trunks” followed by “Trunk Groups” within ShoreWare Director. 1. Open the 'ShoreWare Director Admin' page click on (Administration > Trunks > Trunk

Groups): This action brings up the “Trunk Group” screen

1. From the pull down menus on the “Trunk Groups” screen, select the site desired and select the “SIP” trunk type to configure and click on the “Go” link from “Add new trunk group at site:”. The “Edit SIP Trunk Group” screen will appear.

SIP Trunk Group Settings 1. Within the “Edit SIP Trunks Group” screen define a name for the trunk group, in this

example we chose “AudioCodes / Optus”. 2. The next step is to verify the setting of the “Teleworker” check box. The “Teleworker”

check box needs to be disabled (unchecked), this is the default setting. 3. "The "Profile:" parameter is a drop down selection, change from the default setting of

"_SystemTrunk" to a setting of "_ATTBVOIP"" 4. The “Enable Digest Authentication” field is not required when connecting to an

AudioCodes device. 5. The “Enable SIP Info for G.711 DTMF Signaling” box should not be checked. Enabling

SIP info is currently only used with tie trunks between ShoreTel systems. 6. The next item to change in the “Edit SIP Trunks Group” screen is to make the

appropriate settings for the “Inbound:” fields. 7. Within the “Inbound:” settings ensure the “Number of Digits from CO” is set to 10 for

Optus and ensure the “DNIS” or “DID” box is checked, along with the Extension parameter (see Planning and Installation Guide for further information on configuration).

8. Enable the “Tandem Trunking” parameter, this will allow the ShoreTel system to transfer calls to external parties via the SIP trunks. Select an appropriate “User Group” that has access to this “Trunk Group” being configured.


9. The last item to define is the “Destination” parameter, this will determine where an inbound call is routed if there isn’t a DNIS, DID or extension match, we chose the default Auto Attendant menu.

Outbound and Trunk Services 1. If this trunk group is to be used for outbound calls, be sure to enable (check) the

“Outbound:” parameter, then define an “Access Code:” and “Local Area Code:”. Please refer to the ShoreTel Planning and Installation Guide for additional information.

2. The parameter “Caller ID not blocked by default” determines if the call is sent out as <unknown> or with caller information (Caller ID). User DID etc. will impact how information is passed out to the SIP Trunk group, this parameter needs to be enabled (checked).

3. The parameter “Dial Local Numbers in National Form” needs to be enabled to properly

send calls to the Optus network.

4. After these settings are made to the “Edit SIP Trunk Group” screen, press the “Save” button to save the changes. This completes the SIP Trunk Group configuration.


If this trunk group is to be used for outbound calls, be sure to enable (check) the

“Outbound:” parameter, then define an “Access Code:” and “Local Area Code:”. Please

refer to the ShoreTel Planning and Installation Guide for additional information.


3.2.5 System Settings – Individual Trunks

This section covers the configuration of the individual trunks. Select “Administration”, then “Trunks” followed by “Individual Trunks” to configure the individual trunks. Individual Trunks

1. Open the 'ShoreWare Director Admin' page click on (Administration > Trunks > Individual Trunks): This action brings up the “Individual Trunks” screen

2. The “Trunks by Group” screen that is used to change the individual trunks settings then appears.

Trunks by Group

3. Select the site for the new individual trunk(s) to be added and select the appropriate trunk group from the pull down menu in the “Add new trunk at site” area. In this example, the site is “Australia” and the trunk group is “AudioCodes / Optus”, which was created above. Click on the ”Go” link to bring up the “Edit Trunk” screen.


Edit Trunks Screen for Individual Trunks

4. From the individual trunks “Edit Trunk” screen, input a name for the individual trunks, in this example “Optus” was used. When selecting a name, the “best practice” is to name the individual trunks the same as the name of the trunk group so that the trunk type can easily be tracked. Select the switch upon which the individual trunk will be created.

5. Select the appropriate switch.

6. In the “IP Address:” parameter input the IP address of the AudioCodes LAN interface (usually eth0). The last step is to select the number of individual trunks desired (each one supports “one” audio path – example if 30 is define, then 30 audio paths can be up at one time). Once these changes are complete, press the “Save” button to input the changes.

Note: Individual SIP Trunks cannot span networks. SIP Trunks can only terminate on the switch selected. There is no failover to another switch. For redundancy, two trunk groups will be needed with each pointing to another AudioCodes device – just the same as if PRI were being used. After setting up the trunk groups and individual trunks, refer to the ShoreTel Product Installation Guide to make the appropriate changes for the User Group settings. This completes the settings for the ShoreTel system side


4 AudioCodes AudioCodes Ltd. designs, manufactures and sells advanced Voice over IP and converged VoIP and Data networking products and applications to Service Providers and Enterprises. AudioCodes’ products are deployed globally in IP, Mobile, Cable, and Broadband Access networks, as well as small, medium and large Enterprises. The company provides a diverse range of innovative, cost-effective products for converged VoIP and Data networks including Media Gateways, Enterprise Session Border Controllers (E-SBC), Residential Gateways, Multi-Service Business Gateways, IP Phones, Mobile VoIP Clients, Media Servers and Value Added Applications.

4.1 AudioCodes Enterprise SBC (E-SBC) Product Information

AudioCodes' family of Enterprise Session Border Controllers enables connectivity and security between enterprises and Service Providers’ VoIP networks. The E-SBC family provides Perimeter Defense as a way of protecting companies from malicious VoIP attacks; mediation for allowing the connection of any PBX and/or IP-PBX to any Service Provider; and Service Assurance for service quality and manageability. The native implementation of SBC functions on the AudioCodes Mediant Media Gateways and Multi-Service Business Gateways provides a host of additional capabilities that are not possible with standalone SBC appliances, such as VoIP mediation, PSTN Access, data routing, WAN access, data security, survivability, and third party value-added services applications. This enables enterprises to utilize the advantages of converged networks and eliminate the need for standalone appliances.

4.2 AudioCodes Product Configuration The following section describe the configuration of the AudioCodes eSBC products.

Web Interface Basics for the Mediant 1KB-MSBG Configuration basics When using the browser to configure the Mediant, you need to be aware of a few useful tips and icons. IMPORTANT: Always use the Full tree when programming

When you change a parameter in a multi-parameter table, an icon will appear beside the changed value to indicate it has to be written to the system.


Clicking the “Submit” or OK button will write the changes.

When you are logged onto the system you can click the “Burn” button at the top of the screen. This writes the configuration to flash and ensures that your programming will not be lost if power is removed from the unit. NOTE: It is advisable to use this after every config change. Some changes require a reset of the system and when you make changes of this type you get an indication at the top of the screen

Follow the same procedure as in the AudioCodes Mediant 1KB-MSBG Prerequisites area for resetting the Mediant

4.3 Configuration of the AudioCodes eSBC products The following section describe the configuration of the AudioCodes eSBC products.

To configure the LAN IP interface:

1. Open the Multiple Interface table (Configuration tab > VoIP menu > Network submenu > IP Settings). Assign the Application Type, IP address, Prefix Length/Subnet Mask, Gateway, VLAN ID, Interface Name and DNS Server. Take note that for the Interface Name the word “WAN” should NOT be used as it is a reserved word. The Interface Name can be chosen arbitrarily.

2. Add the IP interface entry for the network signaling leg:

a. Application Type = OAMP + Media + Control

b. IP address = Enter LAN IP Address

c. Prefix Length = 24

d. Gateway = Enter Default Gateway IP Address

e. VLAN ID = 1

f. Interface Name = Voice

g. Primary DNS Server IP Address = Enter DNS IP Address

h. Secondary DNS Server IP Address = Enter Secondary DNS IP Address

i. Click Apply, and then done to validate the entry.


Figure 4-1: Defining the LAN Interface IP Address

3. Save the settings to flash memory ("burn") and reset the device.

To configure the LAN Switch Vlan 1 interface:

This interface is will route between LAN and WAN interfaces: The IP address assigned should be the same as the IP address assigned to gateway filed in the ‘Multiply Interface’ Table

4. Open connection page (Configuration >> Data >> Data System >> Connections) Select edit icon next to ‘LAN switch Vlan1’ parameter, select Setting tab then enter the IP address

Figure 4-2: Defining the Internal LAN switch Vlan 1 IP Address

Make sure IP address corresponds to gateway IP address in the Multiply Interface table (see point 1)


To configure the WAN interface:

5. Open ‘WAN access setting Page‘(Configuration >> Data >> WAN Access >> Setting) and configure the flowing

a. Connection Type: Manual IP address Connection

b. IP address: Enter WAN IP Address

c. Netmask : Enter Netmask

d. Default gateway: Enter DG IP Address

e. Primary DNS : Enter WAN DNS If required

Figure 4-3: Defining the WAN IP Address

Use the drop down box and select the Manual IP Address Ethernet Connection

Enter the required IP information supplied by Optus (eg. WAN IP Address, subnet,

gateway and primary DNS)

Click “Apply” to activate the changes.

NOTE: Don’t forget to “Burn” as you go to save your config to flash memory in the unit.


Configure NAT if required:

6. NAT setup (only if required) (Configuration >> VoIP >> SIP Definitions >> General

Parameters)

Configure Routing Table:

7. NAT setup (only if required) (Configuration >> VoIP >> Network >> IP Routing Table)


Enable SBC Application

8. In order to configure the rest of the gateway the SBC feature of the Mediant needs to be enabled using the following steps. (The license and hardware is covered in the pre-requisite area of this document.)

(Configuration >> VoIP >> Application Enabling)

Using the drop down box select Enable for the SBC option. NOTE: As this change requires a reset, perform a reset as described in the Auxiliary Files area of this document.

IP Media Settings Table

9. In order to ensure there are Media channels available for the transcoding required between Optus and the ShoreTel Configure the IP Media settings using the following steps.

(Configuration >> VoIP >> Media >> IPMedia)

• Assign the value to Number of media channels


Change the Number of Media Channels to the number required for the deployment (up to 120 giving 60 simultaneous calls) this is also based on the hardware and licenses discussed previously in this guide.

Click Submit to save the change in the table. NOTE: As this change requires a reset, perform a reset as described below.

Reboot Device

10. Perform a reset as described below.

(Maintenance >> Maintenance Actions >> Reset)

1. SIP Media Realm Table

SIP Media Realm

11. Media Realms are used to separate parts of the gateway to allow different signalling and codec support (AudioCodes likens it to configuring virtual gateways each with its own characteristics). Media Realms are used in our instance to allow internal connections to be presented as public facing connections by transcoding information between the two realms. To configure media realms use the following steps

(Configuration >>VoIP>>Media>>Media Realm Configurations >> Add)


This Opens the Media Realm Window

Lan Realm

Set media name: LANREALM

IPv4 interface : LAN

IPv6 interface : none

Port range start (valid range 6000-9000)

Number of Media Session legs : e.g. 30 sessions

Click on Submit

Wan Realm

Set media name: WANREALM

IPv4 interface : LAN

IPv6 interface : none

Port range start (valid range 6000-9000 different then LAN )

Number of Media Session legs :

Click on Submit


Reboot Device




SRD Table

13. SRD’s are Signaling Routing Domains. These are used by each media realm to define the signaling characteristics of the Realm. SRD’s can be configured for this solution using the following steps.

(Configuration >> VoIP >> Control Network >> SRD Table)

LAN SRD

Select Index 1 = 1 - LANSRD

SRD name : LANSRD

Media Realm : LANREALM

WAN SRD

Select Index 2 = 2 - WANSRD

SRD name : WANSRD

Media Realm : WANREALM


SIP Interface Table

14. As each signalling realm is associated with a “virtual gateway” via a Media Realm it

needs a specific SIP Interface with unique settings Each interface will have a specific port associated so each “virtual gateway” does not interfere with another. To set up the SIP interfaces use the following steps. (Configuration >> VoIP >> Control Network >> SIP Interface Table >> Add)

Create LAN and WAN interface and assign corresponding SRD

Choose the name of the Network Interface to be associated with the SIP Interface.

LAN SIP Interface

Network Interface = Enter LAN for the first Interface

Using the drop down box choose SBC as the Application Type

As the gateway is required to use port 5060 for the Optus connection the ports for the PBX (internal side of the gateway) need to be changed

Use 5060 for the UDP and TCP ports and 5061 as the TLS port

This interface is being associated with the LAN side of the gateway so enter 1 in the SRD field

Click Apply to save the entry in the table

Now enter 1 in the Add field and click the Add button


WAN SIP Interface

Network Interface = Enter WAN for the second Interface

Using the drop down box choose SBC as the Application Type

As the gateway is required to use port 5060 for the Optus connection the ports for the PBX (internal side of the gateway) need to be changed

Use 5060 for the UDP and TCP ports and 5061 as the TLS port

This interface is being associated with the WAN side of the gateway so enter 2 in the SRD field

Click Apply to save the entry in the table

NOTE: Don’t forget to “Burn” as you go to save your config to flash memory in the unit.

Proxy Sets Table

15. We need to define the proxies used for the interconnection of the devices internally and

externally. Using the Proxy Sets Table also allows us to define “keep alive” options to ensure the connectivity is active.

To configure the Proxy Sets ID use the following steps.

(CONFIGURATION >>VOIP>>CONTROL NETWORK>>PROXY SET TABLE)

This opens the Proxy Sets window

Using the drop down box select Proxy Set ID 1 (ID 0 is ONLY used when a default proxy is configured in the gateway)

ShoreTel Proxy – index1


Enter the IP address of the PBX (eg 10.195.0.45)

Choose the Transport Type required by the PBX (eg UDP)

Assign the Proxy Set to the relevant SRD (eg 1)

Click Submit to update the table.

Using the drop down box select Proxy Set ID 2

Configure the following settings

Optus Proxy – Index 2

Using the drop down box select Proxy Set ID 2

Enter the remote IP address supplied by Optus

Choose the Transport Type required by the PBX (eg UDP)

Assign the Proxy Set to the relevant SRD (eg 2)

Click Submit to update the table.

Reboot Device




IP Profiles

17. Open IP Profile settings (CONFIGURATION >>VOIP>>CODERS AND PROFILES>>IP PROFILE SETING)

This will open the IP Profiles settings window

Profile ID 1- ShoreTel

Select Profile ID 1

In the first section give the Profile a Name (eg ShoreTel)


Change the Disconnect on Broken Connection option to Yes.

Change the Enable Early Media option to Enable.

Profile ID 2 – OPTUS

Select Profile ID 2

In the first section give the Profile a Name (eg Optus SIP Evolve)

Change the Disconnect on Broken Connection option to Yes.

Change the Enable Early Media option to Enable.

IPGroup Table

18. IP Groups are used to assign various settings to realms and devices.


To setup IP Groups use the following steps. (CONFIGURATION >>VOIP>>CONTROL NETWORK>>IP GROUP TABLE)

This opens the IP Groups Table window Using the drop down box select Index 1


Index 1 - ShoreTel Select Index 1

Type: Server

Description: ShoreTel 12.x

Proxy set ID : 1

SIP Group Name : PABX IP address

Media Realm : LANREALM

IP profile ID : 1

Click Submit to save the settings in the group.

Index 2 - Optus Select Index 2

Type: Server

Description: Optus

Proxy set ID : 2

SIP Group Name : Optus proxy IP address

Media Realm : WANREALM

IP profile ID : 2

Click Submit to save the settings in the group.


Classification Table

19. CONFIGURATION >>VOIP>>CONTROL NETWORK>>SBC >>ROUTING SBC >>Add

Index 1

Source SRD ID = 1

Source IP Address = IP Address of ShoreTel SIP Trunk IP’s

Source IP Group ID =1

Index 2

Source SRD ID = 2

Source IP Address = IP Address of Optus Evolve SIP Trunks

Source IP Group ID =2

IP2IP Routing Table

20. The two realms of the SBC need to be configured to route calls between each other. To

configure this use the following steps

(CONFIGURATION >>VOIP>>SBC>>ROUTING SBC>>IP TO IP ROUTING TABLE)

This opens the IP2IP Routing Table window.


Type 1 in the Add field and click the Add button

Configure the first (1) index entry as below

Change the Source IP Group ID to 1 (this associates this entry with calls coming from the PBX)

Change the Destination IP Group ID to 2

Change the Destination SRD ID to 2

These changes associate the routing of PBX calls towards the Optus SIP Trunks.

Click Apply to save the changes in the table.

Configure the Second (2) index entry as below

Change the Source IP Group ID to 2 (this associates this entry with calls coming from Optus SIP Trunks)

Change the Destination IP Group ID to 1

Change the Destination SRD ID to 1

These changes associate the routing of Optus SIP Trunks towards the ShoreTel SIP Trunks.

Click Apply to save the changes in the table.


Reboot Device




Backing up the Config file

22. (Maintenance >> Software Update >> Configuration File)

This will open the Configuration File window

Click the Save INI File in the top section (this is the voice gateway part of the

unit)

Click on the save button.


Document and Software Copyrights Copyright © 2012 by ShoreTel, Inc., Sunnyvale, California, U.S.A. All rights reserved. Printed in the United States of America. Contents of this publication may not be reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose, without prior written authorization of ShoreTel Communications, Inc. ShoreTel, Inc. reserves the right to make changes without notice to the specifications and materials contained herein and shall not be responsible for any damage (including consequential) caused by reliance on the materials presented, including, but not limited to typographical, arithmetic or listing errors.

5 Trademarks The ShoreTel logo, ShoreTel, ShoreCare, ShoreGear, ShoreWare and ControlPoint are registered trademarks of ShoreTel, Inc. in the United States and/or other countries. ShorePhone is a trademark of ShoreTel, Inc. in the United States and/or other countries. All other copyrights and trademarks herein are the property of their respective owners.

6 Disclaimer ShoreTel tests and validates the interoperability of the Member's solution with ShoreTel's published software interfaces. ShoreTel does not test, nor vouch for the Member's development and/or quality assurance process, nor the overall feature functionality of the Member's solution(s). ShoreTel does not test the Member's solution under load or assess the scalability of the Member's solution. It is the responsibility of the Member to ensure their solution is current with ShoreTel's published interfaces. The ShoreTel Technical Support organization will provide Customers with support of ShoreTel's published software interfaces. This does not imply any support for the Member's solution directly. Customers or reseller partners will need to work directly with the Member to obtain support for their solution.

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