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-13053 Date: August, 2013 Product: ShoreTel Audiocodes Optus System version: ShoreTel 13.x Application Note ShoreTel / Audiocodes / Optus SIP Trunking 28 August 2013 Version 2 Issue 1 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.
Table of Contents APPLICATION NOTE... 1 SHORETEL / AUDIOCODES / OPTUS SIP TRUNKING... 1 1 OVERVIEW... 3 1.1 DOCUMENT FEEDBACK... 4 1.2 SIP TRUNKING ARCHITECTURE OVERVIEW... 4 2 INTEROPERABILITY REQUIREMENTS, VALIDATION AND LIMITATIONS... 4 2.1 VERSION SUPPORT... 5 2.2 LIST OF COMPLIANT HARDWARE... 5 2.2.1 ShoreTel System... 5 2.2.2 Audiocodes... 5 2.2.3 Voice switches... 5 2.2.4 Phones... 5 2.3 VALIDATION RESULTS SUMMARY... 6 3 SHORETEL... 12 3.1 SHORETEL UNSUPPORTED FEATURES... 12 3.2 SHORETEL CONFIGURATION... 12 3.2.1 Call Control Settings... 13 3.2.2 Sites Settings... 15 3.2.3 Switch Settings - Allocating Ports... 17 3.2.4 System Settings Trunk Groups... 19 3.2.5 System Settings Individual Trunks... 22 4 AUDIOCODES... 24 4.1 AUDIOCODES ENTERPRISE SBC (E-SBC) PRODUCT INFORMATION... 24 4.2 AUDIOCODES PRODUCT CONFIGURATION... 24 4.3 CONFIGURATION OF THE AUDIOCODES ESBC PRODUCTS... 26 DOCUMENT AND SOFTWARE COPYRIGHTS... 47 5 TRADEMARKS... 47 6 DISCLAIMER... 47 7 COMPANY INFORMATION... 47
1 Overview This document provides details for connecting the ShoreTel UC system though the Audiocodes Mediant gateway products to Optus for SIP Trunking 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 is connected to both the LAN and WAN and acts as a secure gateway 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 offered by Optus Australia in a secure manner is important. The Audiocodes SBC then 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 validated ITSPs
1.1 Document Feedback ShoreTel IP PBX administrators who would like to provide feedback on the contents of this document should send it to TPPfeedback@shoretel.com. 1.2 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: ShoreTel SIP Trunks 10.195.0.45 Optus SIP Trunks xxx.xxx.xxx.131 Optus Evolve SIP Trunks AudioCodes LAN IP Address 10.195.0.44 Optus WAN IP Address xxx.xxx.xxx.115 2 Interoperability Requirements, Validation and Limitations
2.1 Version Support Products are Validated via the ShoreTel Innovation Network Validation Process for the ShoreTel system. The table below contains the matrix of Audiocodes versions firmware releases validated on the identified ShoreTel software releases. Audiocodes esbc version 6.40A.050.004 ShoreTel 13.x 2.2 List of compliant hardware The following hardware while not necessarily used during the validation itself is guaranteed to comply with the testing results of this application note. 2.2.1 ShoreTel System The ShoreTel system used during the validation process is a ShoreWare Director Enterprise Edition Server consisting of ShoreTel 13.0 version. A ShoreTel Small Business Edition Server is also compatible with Optus Australia SIP Trunks. 2.2.2 Audiocodes The Audiocodes E-SBC system used during validation is a Mediant 800, 1000 & 2000 esbc consisting of Firmware version 6.40A.016.003 AudioCodes Mediant 1000 Enterprise Session Border Controller (E-SBC) is a member of AudioCodes family of Enterprise Session Border Controllers, enabling connectivity and security between enterprises and Service Providers VoIP networks. AudioCodes Mediant 1000 E-SBC is 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 Denail 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). 2.2.3 Voice switches ShoreGear 30 ShoreGear 50 ShoreGear 50v ShoreGear 90 ShoreGear 90v ShoreGear 90BRI ShoreGear 90BRIv ShoreGear 120 ShoreGear 220T1 ShoreGear 220T1a ShoreGear 220E1 2.2.4 Phones Regardless the technology of phones connected to the ShoreTel system, all officially supported phones may place calls over an Optus SIP trunk & will demonstrate the same behavior (features & limitations) as captured in this application note.
2.2.4.1 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 2.2.4.2 SIP phones All officially ShoreTel Innovation Network Validated SIP phones are also compliant with the test results documented in this application note. 2.2.4.3 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 2.3 Validation Results Summary Test Plan for Optus SIP Trunking Service Table 2-1: Initialization and Basic Calls
ID Name Description Notes 1.0 Configuration The Innovation Network Lab will use the configuration Application Note application note provided by the vendor to configure the vendor s product to work with the ShoreTel system. 1.1 Setup and Verify successful setup and initialization of the SUT initialization 1.2 Outbound Call (Domestic) Verify calls outbound placed through the SUT reach the external destination. see Note 1 1.3 Inbound Call (Domestic) Verify calls received by the SUT are routed to the default trunk group destination. see Note 1 1.4 Device restart Verify that the SUT recovers after power loss to the Power Loss SUT 1.5 Device restart Verify the SUT recovers after loss of network link to Network Loss the SUT. 1.6 All Trunks Busy Verify an inbound callers hears busy tone when all Inbound Callers channels/trunks are in use 1.7 All Trunks Busy Verify an outbound callers hears busy tone when all Outbound Callers 1.8 Incomplete Inbound Calls channels/trunks are in use Verify proper call progress tones are provided and proper call teardown for incomplete inbound calls. Table 2-2: Media and DTMF Support ID Name Description Notes Erro r! Refe renc e sour ce not foun d. Error! Reference source not found. Error! Reference source not found. 2.2 Media Support SIP Reference to SUT 2.3 Codec Negotiation 2.4 DTMF Transmission Out of Band / In Band 2.5 Auto Attendant Menu 2.6 Auto Attendant Menu Dial by Name 2.7 Auto Attendant Menu checking Voice Mail mailbox Verify call connection and audio path from a SIP Reference phones to an external destination through the service provider using all supported codes with both sides set to a common codec. Verify codec negotiation between the SUT and the calling device with each side configured for a different codec. Verify transmission of in-band and out-of-band digits per RFC 2833 for various devices connected to the SUT. Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the desired extension. 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. Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the Voice Mail Login Extension. See Note 2
Table 2-3: Performance & Quality of Service ID Name Description Notes 3.1 Voice Quality Verify the SUT can provide a voice quality SLA across Not Tested Service Levels the WAN from the customer premises to the SUT SIP gateway. 3.2 Capacity Test Verify the service provider interface can sustain services through period of heavy outbound and inbound load. 3.3 Post Dial Delay Verify that post dial delay is within acceptable limits. 3.4 Billing Accuracy Verify that all test calls made are accurately reflected in the SUT s CDR and billing reports.
Table 2-4: Enhanced Services and Features ID Name Description Notes 4.1 Caller ID Name and Number - Inbound Verify that Caller ID name and number is received from SIP endpoint device 4.2 Caller ID Name and Number - Outbound Verify that Caller ID name and number is sent from SIP endpoint device 4.3 Hold from SUT to SIP Reference Verify successful hold and resume of connected call 4.4 Call Forward - Verify outbound calls that are being SUT forwarded by the SUT are redirected and connected to the appropriate destination. 4.5 Call Forward Verify outbound calls that are being External PSTN forwarded by the SUT are redirected Number and connected to the appropriate destination. 4.6 Call Transfer Verify a call connected from the SUT blind to the ShoreTel phone can be transferred to an alternate destination. 4.7 Call Transfer Verify a call connected from the SUT Consultative to the ShoreTel phone can be transferred to an alternate destination. 4.8 Conference ad Verify successful ad hoc conference of hoc three parties 4.9 Inbound Verify the SUT provides inbound DID/DNIS dialed number information and is correctly routed to the configured destination. 4.10 Outbound 911 Verify that outbound calls to 911 are (Emergency 000) routed to the correct PSAP for the calling location and that caller ID information is delivered. 4.11 Operator Assisted Verify that 0+ calls are routed to an operator for calling assistance. 4.12 Inbound / Verify that calls with Blocked Caller Outbound call ID route properly and the answering with Blocked phone does not display any Caller ID Caller ID information. 4.13 Inbound call to a Hunt Group 4.14 Inbound call to a Workgroup 4.15 Inbound call to DNIS / DID and leave a voice mail message 4.16 Call Forward FindMe 4.17 Call Forward Always 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. 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. 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. Verify that inbound calls are forwarded to a user s FindMe destination. Verify that inbound calls are immediately automatically forwarded to a user s external destination. See Note 3 See Note 3 See Note 1 Conditional See Note 4 See Note 2
ID Name Description Notes 4.18 Inbound / Outbound Fax calls Verify that inbound / outbound fax calls complete successfully. 4.19 ShoreTel Service Appliance Unified Communication System 4.20 Inbound call to Bridged Call Appearance (BCA) extension 4.21 Inbound call to a Group Pickup extension 4.22 Office Anywhere External Verify that inbound calls are properly forwarded to the ShoreTel Service Appliance and it properly accepts the access code and you re able to participate in the conference bridge. 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. Verify that inbound calls 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 Verify that inbound calls are properly presented to the Office Anywhere External PSTN destination. 4.23 Simul Ring Verify that inbound calls are properly presented to the desired extension and the Additional Phones destinations. 4.24 MakeMe Conference Verify that an inbound call can be conferenced with three (or more) additional parties 4.25 Park / Unpark Verify that an inbound call can be parked and unparked 4.26 Call Recording Verify that external calls can be recorded via the SIP Trunk using ShoreTel Communicator 4.27 Silent Monitor / Barge-In / Whisper Page 4.28 Long Duration Inbound 4.29 Long Duration Outbound Verify that external calls can be silently monitored, barged-in and whisper paged via the SUT. Verify that an inbound call is established for a minimum of 30 minutes. Verify that an outbound call is established for a minimum of 30 minutes. 4.30 Contact Center Verify that an inbound call can be established directly to the ShoreTel Contact Center, that all prompts are heard and the agent can answer the call. Table 2-5: Security ID Name Description Notes 5.1 Registration or Digest Authentication Verify the SUT supports the use of registration or digest authentication for service access for inbound and outbound calls. See Note 5 Fail See Note 6 NA
Note 1: When a user receives external inbound call and tries to Blind Transfer call to another external party using same SIP Trunk, it might fail due to ShoreTel defect 1-112076581 Note 2: DTMF may not work when Full-width Switches (SG 40/8, SG 60/12, SG120/24 etc.) are configured in the System and G.729 codec is used. In this particular scenario, one of the available Full-width Switches will act as a Proxy and will try to do media transcoding. There is a Possibility ShoreTel Server mightn t recognize DTMF due to ShoreTel defect- 1-116701691. This issue will not exist if all of your ShoreTel Switches are Half-Width Switches (SG 50, SG 90 etc.) Note 3: Caller ID will only display Numbers and not Names. Note 4: If Inbound Call goes to WG and no Agent Answers the Call and then call get transferred to 2 nd WG, Call could fail due to ShoreTel defect 1-118194571 Note 5: G.711 pass-through was used for inbound and outbound fax transmissions on Optus network. Note 6: Inbound Call routed to Contact Center might fail due to ShoreTel defect 1-116701925
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. 3.1 ShoreTel Unsupported Features Please refer to the ShoreTel Administration Guide, Chapter 18 Session Initiation Protocol, for supported and unsupported features via SIP Trunks. Following are some feature limitations via SIP Trunks: General Feature Limitations Fax redirect not supported via SIP Trunks using G.711 (though Direct Inward Dialing (DID) to fax endpoint is supported) ShoreTel supports Music On Hold (MOH) over SIP trunks. The maximum number of music on hold (MOH) streams that a SIP-enabled switch can support varies with the switch model. The range of such streams across all the voice switch models is 14 60. Limitation: MOH source needs be on SIP trunk switch. 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. ShoreTel supports the Service Appliance (SA-100) conferencing / IM system from Release -12. SIP trunk calls from / to the SA-100 is supported. The SA- 100 accepts access codes in DTMF RFC2833 only. 4 to 6 party conferences, when a SIP trunk is involved, utilize Make Me conference ports. Silent Monitoring, Barge-In, Silent Coach, Park/Unpark, Call recording features are supported on a SIP trunk call only if SIP trunk configured with Default ITSP SIP profile and the trunk is on a half-width switch. Silence detection on trunk-to-trunk transfers is not supported, it requires a physical trunk. There may be other feature limitations when using SIP Trunks. Please consult the ShoreTel Administration Guide for further details. 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 basically 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) 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. 4) 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). 5) 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. 6) 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 / Server Appliances > Primary): This action brings up the Switches screen 2. From the Switches screen simply select the name of the switch to configure. 3. From the Switches screen simply select the name of the switch to configure.
4. 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. 5. Configure SIP Media Proxy Port on Switch by selecting SIP Trunk with Media Proxy. SIP Trunk Media Proxy is required for Media Hairpinning for following features: Office Anywhere Simulring Trunk Recording Silent Monitor Barge In Silent Coach Silent Monitor For further information on SIP Media Proxy please refer to Chapter 18 of the ShoreTel 13 System Administration Guide.
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 drop down selection, no modification is necessary, leave at a default setting of Default ITSP 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 describes the configuration of the Audiocodes esbc products. Web Interface Basics for the AudioCodes Mediant 800, 1000 & 2000 E-SBC 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 800, 1000 & 2000 E-SBC Prerequisites area for resetting the Mediant
4.3 Configuration of the Audiocodes esbc products The following section describes the configuration of the Audiocodes Mediant esbc products (800, 1000 & 2000). 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 LAN & WAN IP interface entry for the network signaling leg: LAN Interface a. Application Type = OAMP + Media + Control b. Interface Mode = IPv4 Manual c. IP address = Enter LAN IP Address d. Prefix Length = 24 e. Gateway = Enter Default Gateway IP Address f. VLAN ID = 1 g. Interface Name = ShoreTel h. Primary DNS Server IP Address = Enter DNS IP Address i. Secondary DNS Server IP Address = Enter Secondary DNS IP Address j. Underlying Interface = GROUP_1 k. Click Apply, and then done to validate the entry. WAN Interface a. Application Type = Media + Control b. Interface Mode = IPv4 Manual c. IP address = Enter WAN IP Address, Supplied By Optus d. Prefix Length = Supplied by Optus e. Gateway = Enter Default Gateway IP Address, Supplied By Optus f. VLAN ID = 10 Different to LAN VLAN ID g. Interface Name = Optus h. Primary DNS Server IP Address = Enter DNS IP Address, Supplied By Optus i. Secondary DNS Server IP Address = Enter Secondary DNS IP Address, Supplied By Optus j. Underlying Interface = GROUP_2 Figure 4-1: Defining the LAN & WAN Interface IP Address
To confirm physical port allocation : Each pair of ports on M800 by default is allocated to differt Group. To confirm what ports are in Group 1 and Group2 as defined in the network Interface setting: 1. Open connection page (Configuration tab >>Voip menu >Network submenu >>Phisical port setting to confirm what ports are in Group 1 and Group 2 as defined in the network Interface setting : Figure 4-2: Phisical port setting NOTE: In case different VLNA ID has to be assigned to each interface (if it same interface type- IPv4) For example: Interface Voice VLAN ID 1 Group 1 Interface External VLANID 2 Group 2 Make sure that the ports allocated in the Physical port setting have corresponding VLAN ID For example Group 1 VLANID 1 Group 2 VLANID 2 Otherwise the packets will be tagged with the VLAN ID specified in the multiply interface table 3) Save the settings to flash memory ("burn") and reset the device. Configure Routing Table: 6. Open the Multiple Interface table (Configuration tab > VoIP menu > Network submenu > IP Routing Table).
Configure RFC 2833 Payload Type Table: 7. Open the Multiple Interface table (Configuration tab > VoIP menu > Media submenu > RTP/RTCP Settings). 8. The RFC 2833 TX & RX Payload Type parameter defaults to a value of 96, you will need to ensure that you change this parameter to a value of 101 for both the TX and RX Payload Type to interoperate with Optus. Enable SBC Application 9. 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 prerequisite 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 10. 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, this is based on the type of AudioCodes E-SBC deployed and licenses purchased. Click Submit to save the change in the table. NOTE: As this change requires a reset, perform a reset as described below. Reboot Device 11. Perform a reset as described below.
(Maintenance >> Maintenance Actions >> Reset) SIP Media Realm 12. 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 LANREALM Set media name: LANREALM IPv4 interface : ShoreTel IPv6 interface : none Port range start (valid range 6000-9000) Number of Media Session legs : e.g. 30 sessions Click on Submit WANREALM Set media name: WANREALM IPv4 interface : Optus IPv6 interface : none Port range start (valid range 6000-9000 different then LANREALM ) Number of Media Session legs : Click on Submit
Reboot Device 13. Perform a reset as described below. (Maintenance >> Maintenance Actions >> Reset) SRD Table 14. 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 15. As each signaling 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 ShoreTel 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 Message Policy enter None 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 Optus 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 Message Policy enter None 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 16. 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 ShoreGear Switch providing the SIP Trunks (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 17. Perform a reset as described below. (Maintenance >> Maintenance Actions >> Reset) Configure SIP Definitions: 18. NAT setup (only if required) (Configuration >> VoIP >> SIP Definitions >> General Parameters) 19. PRACK Mode = Supported 20. Enable Early Media = Enable
IP Profiles 21. Open IP Profile settings (CONFIGURATION >>VOIP>>CODERS AND PROFILES>>Coders) 22. Enable G.711 A-Law 23. Enable G.711 U-Law 24. Enable G.729 25. 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 26. 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 27. 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 Port = 0 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 28. 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.
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 29. Perform a reset as described below. (Maintenance >> Maintenance Actions >> Reset)
Backing up the Config file 30. (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. 7 Company Information ShoreTel, Inc. 960 Stewart Drive Sunnyvale, California 94085 USA +1.408.331.3300 +1.408.331.3333 fax