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Document Overview

This document provides a configuration guide for Sonus SBC 5XX0 Series (Session Border Controller) when connecting to Genesys Voice Platform.

This configuration guide supports features documented in the Genesys Voice Platform guides.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flows between Sonus SBC 5XX0 and Genesys Voice Platform.

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. There will be steps that require navigating third-party as well as the Sonus SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary.

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate, but are presented without warranty of any kind, express or implied, and are provided "AS IS." Users must take full responsibility for the application of the specifications and information in this guide.


Requirements

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

 

Table : Requirements

 

Equipment

Software Version

Sonus Networks 

Sonus SBC 5200
BMC
BIOS
ConnexIP OS
SonusDB
EMA
SBX

V05.01.01-R000
V02.16.00
V02.06.00
V03.01.01-R000
V05.01.01-R000
V05.01.01-R000
V05.01.01-R000

Third-party Equipment



Genesys SIP Server8.1.100.98
Genesys GVP MCP

8.1.504.93

Polycom SoundPoint IP 501 SIP

2.1.3


Reference Configuration

The following reference configuration shows connectivity between third-party and Sonus SBC 5XX0.

Figure : Connectivity Between Third-Party and Sonus SBC 5XX0

Support

For any questions regarding this document or the content herein, please contact your maintenance and support provider.

 

Third-Party Product Features

The testing was executed with the Genesys test plan. The following features were tested:

Inbound calls 

  • Inbound Call to Agent released by caller
  • Inbound Call to Agent released by agent
  • Inbound Calls rejected
  • Inbound Call abandoned
  • Inbound Call to Route Point with Treatment
  • Interruptible Treatment
  • IVR (Collect Digit) Treatment
  • Inbound Call routed by using 302 out of SIP Server signaling path

Outbound Calls

  • 1PCC Outbound Call from SIP Endpoint to external destination
  • 3PCC Outbound Call to external destination
  • 1PCC Outbound Call Abandoned

Music on hold

  • Caller is put on hold and retrieved by using RFC 2543 method
  • T-Lib-Initiated Hold/Retrieve Call with MOH using RFC 3264 method

Call Transfers

  • 3PCC 2 Step Transfer to internal destination by using re-INVITE method
  • 3PCC Alternate from consult call to main call
  • 1PCC Unattended (Blind) transfer using REFER
  • 1PCC Attended Transfer to external destination
  • 3PCC Two Step Conference to external party
  • 3PCC (same as 1PCC) Single-Step Transfer to another agent
  • 3PCC Single Step Transfer to external destination using REFER
  • 3PCC Single Step Transfer to internal busy destination using REFER

Other scenarios

  • Early Media for Inbound Call to Route Point with Treatment
  • Early Media for Inbound Call with Early Media for Routed to Agent
  • Inbound call routed outbound (Remote Agent) using INVITE without SDP
  • Call Progress Detection
  • Out of Service detection. Checking MGW live status
  • SIP Authentication for outbound calls
  • SIP Authentication for incoming calls

Remote Agent behind SBC scenarios

  • T-Lib-Initiated Answer/Hold/Retrieve Call for Remote SIP endpoint which supports the BroadSoft SIP Extension Event Package
  • 3PCC Outbound Call from Remote SIP endpoint to external destination
  • 3PCC 2 Step Transfer from Remote SIP endpoint to internal destination
  • 1PCC Attended Transfer from Remote SIP endpoint to external destination

Verify License

SBC-POL-RTU

 

 

Genesys Voice Platform Configuration

The following new configurations are included in this section:

  1. Accessing Genesys Tools and Interfaces
  2. Creating SIP Switch in Genesys Administrator
  3. SIP Server Configuration in Genesys Administrator
  4. Genesys Media Server Deployment
  5. Stat Server Configuration

1. Accessing Genesys Tools and Interfaces

This section provides the configuration required for the Genesys components.

Genesys is configured using several different tools and interfaces. The tools and interfaces used in this document are shown below to include their location and method of access.

To access these items, a Remote Desktop Connection (RDC) to the Genesys server is required. The username, password, and IP address of the system to be accessed should be provided by the person(s) installing the Genesys system.

Once logged onto the Genesys system, click the Start button and look for the installed applications shown below. If the applications are not visible on the Start Menu, find them using the search box just above the Start button.

  •  The Solution Control Interface is used to start/stop the various applications, as well as identify the configuration of each application.
  •  The Configuration Manager is a tool used to configure and verify the many settings on the different applications.
  •  The Interaction Routing Designer is used to create and configure Route Points and strategies.
  •  Internet Explorer (not shown in the startup menu) is used to access the Genesys Administrator. The URI should be available from those who installed the platform.

 

Steps can be performed using multiple tools. For example, starting or stopping an application can be performed in the Genesys Administrator as well as the Solution Control Interface.

Figure : Access Solution Control Interface

  • Below is a screenshot of the Solution Control Interface. In this application, click View, top left, to gain access to the area of interest. Click Applications and expand the folders of interest. In this example, SIPServer1 properties are displayed. Clicking the various links will display the appropriate property windows. You can Start/Stop/GracefulStop any application from within this tool.

Figure : Solution Control Interface

  • The Configuration Manager is used to configure the platform and its applications. Once opened, click Applications/Media (it’s possible that your SIP Server is in a different folder under Applications) and the upper left pane will display both SIP Servers. Double-click SIPServer1 and a dialog box will open. To view all options for this server, click the Options tab, and then click the TServer section.

Figure : Configuration Manager

Note that there are many option parameters. Type the name of the option in the filter "notice that it filters in real time". Some options can be set at both Application and Switch/DN levels. The option setting at the DN level takes precedence over the Application-level setting. See the Genesys SIP Server Deployment Guide for details. 

Figure : SIP Server Properties

  • The Interaction Routing Designer is a tool used to create Route Points and/or Strategies. Access the Strategies by clicking Routing Design>Strategies. Double-clicking any strategy will bring up a second window (not shown here). This second window is where Strategies can be created and modified.

Figure : Interaction Routing Designer

  • Access the Genesys Administrator with a web browser. Contact the administrator or person(s) who performed the install of the system to determine the URL. Once opened, click the Provisioning tab and under the Navigation area click Switching. Under Switching, click Switches to display the names of the Switch objects.

Figure : Genesys Administrator - Switches

  • Double-click the switch name and then click the DNs tab. The DNs for your SIP Switch will be shown. Each folder can be double-clicked to access the contents. Underlined are the bread crumbs for navigation. Circled is the icon area to add new DNs or delete existing ones.

Figure : Genesys Administrator - DNs

 


2. Creating SIP Switch in Genesys Administrator

  • Within Genesys Administrator, create a Switching Office named SIPServer Switching Office.

Figure : Genesys Administrator - Creating SIP Switch


  • Within Genesys Administrator, create a SIP Server Switch and associate the Switching Office created in the previous step with this switch.

Figure : Genesys Administrator - SIP Switch Association


  • Under the SIPSwitch created in the above step, define Routing Points to run URS strategies from, the SIP trunk representing connection of SIP Server to Sonus, and a “msml” VoIP service DN required to integrate SIP Server with Media Server to support call hold and conferencing functionalities. 

Figure : Genesys Administrator - Define Routing Points


  • Define DNs of type Extension under SIPSwitch with the following options in the TServer section for various SIP endpoints that will register to the SIP Server.
    • use-contact-as-dn=true Specifies whether SIP Server will use the username of the Contact header as ThisDN.
    • contact=* – Specifies the contact address of the extension DN to which SIP Server should send the SIP call. Here the Contact option value is the IP address of the internal interface of Sonus through which the SIP REGISTER message was received by SIP Server.
    • cpn=<2086041001> SIP Server uses the value of this option as the user part of the SIP URI in the From header of the INVITE message that it sends from this DN to the destination DN. Since this option is used to provide customized caller-ID information to the destination, this option must be configured in the originating DN.
    • sip-cti-control=talk,hold The SIP method NOTIFY (event talk) or NOTIFY (event hold) is used to request the end point to answer or place a call on hold, respectively.


Figure : Genesys Administrator - Extension Options


  • Define a SIP trunk DN to represent all SIP calls arriving from the Sonus SBC internal interface to the SIP Server. Configure the following options under the TServer section of the Trunk DNs.

    Trunk DN:

    • contact=<10.35.141.52:5060> IP address and TCP/UDP port number of the SIP Signaling Port of the Sonus SBC 5000 configured for Genesys. The SIP Signaling Port IP address is used by SIP Server to route or receive calls from test phones through this interface.
    • cpd-capability=mediaserver – Specifies whether SIP Server will use the username of the Contact header as ThisDN.
    • dial-plan=DialPlanInbound – Specifies which dial-plan DN will be applied to calls
    • prefix=<214340> – (NPANXX) Specifies the contact address of the extension DN to which SIP Server should send the SIP call. Here the Contact option value is the IP address of the SIP Signaling Port of the Sonus SBC 1000/2000 through which the SIP REGISTER message was received by SIP Server.

Figure : Genesys Administrator - SIP Trunk Options

  • Define an MSML voice over IP service DN with the following options in the TServer section:
    • contact-list=<IP Address:Port> SIP IP address and listening port for Resource Manager.
    • oos-check=15 Specifies how often (in seconds) SIP Server checks a device for out-of-service status.
    • oos-force=20 Specifies the time interval (in seconds) that SIP Server waits before placing a device that does not respond in out-of-service state when the oos-check option is enabled.
    • prefix=msml= Required for conference and monitoring services only.
    • service-type=msml Specifies the configured SIP device type or service.
    • subscription-id=Resources Specifies the type of subscription ID.

Figure : Genesys Administrator - Define MSML

  • Verify GVP_RM Pair settings:

    • prefix=msml=Required for conference and monitoring services only.

    • refer-enabled=false Specifies the configured SIP device type or service.

    • ring-tone-on-make-call=false Affects the TMakeCall request when using the re-INVITE procedure. When the ring-tone-on-make-call option is set to false, there is no ring tone.

    • make-call-rfc3725-flow=1 Setting this option to 1 instructs SIP Server to use the 3pcc call flow as defined in the RFC 3725.

Figure : Genesys Administrator - GVP_RM Pair Settings

  • Create DNs of type Routing Point in the SIPSwitch which should match the Request URI user part. In this instance it was extensions 1011-1015.

Figure : Genesys Administrator - Create DN of Type Routing Point

 

Figure : Genesys Administrator - Routing Strategies

  • The SIP Server must have Full Control permission for the DN objects under the SIP Server Switch, in order to update various configuration objects under it, such as the Extension DNs.

By default, it does not have this permission. You must grant “Full Control” permission for the System account for the all DNs on the corresponding switch. It is done for all DNs at once by changing the permissions for the system account on the DN folder in the switch object. Or, you can start SIP Server under another account that has change permission on the necessary DNs.

With this full control access, the SIP Server Switch grants DNs like Extension to update their options like “contact” when a new SIP register message is received from end points moving to a new IP location

 

3. SIP Server Configuration in Genesys Administrator

Follow these steps to configure the SIP Server to monitor SIP Server Switch resources, such as SIP extensions/SIP end points registered to SIP Server. The SIP Server also monitors various route points and notifies URS whenever the call arrives on the Route Point.

  • Install and configure SIP Server as per Genesys Framework SIP Server Deployment Guide.
  • Add a connection to the SIP Server Switch created above, to monitor all the resources under this switch: Genesys Administrator-> Provisioning->Environment->Applications->SIP Server Application. Also, SIP Server should add a connection to the tenant. 

Figure : Genesys Administrator - SIP Server Tenant

  • Add a connection to Message Server.

Figure : Genesys Administrator - Add Connection to Message Server

  • Configure the following options in the TServer section in the SIP Server Application object using Genesys Administrator or Configuration Manager:
    • sip-enable-moh=true Enables music on hold.
    • msml-support: true This option and sip-enable-moh (above) allow SIP Server to integrated with Genesys Media server to provide msml/moml-based media services.
    • map-sip-errors=false Genesys Universal Router makes the routing decision for the SIP Server-based solution. If a call fails to route properly, the SIP Server generates an appropriate T-Library error message to inform the router. With this new parameter, SIP Server can now propagate SIP error messages to the router. Setting map-sip-errors=false triggers this functionality in SIP Server.
    • internal-registrar-persistent=true Enables SIP Server to update the DN attribute contact in the configuration database. When an endpoint registers, SIP Server takes the contact information from the REGISTER request and updates or creates a key called contact in the Annex tab of the corresponding DN.
    • sip-dtmf-send-rtp=true In order to support DTMF tone generation on behalf of a 3pcc-based SIP end point application such as  Interaction Workspace SIP end point. When this option is set to true, SIP Server requests Media server to generate RFC 2833 DTMF tones on behalf of the end point.
    • after-routing-timeout Set to 10 seconds. If SIP Server does not get a response on routing a call to SIP agent/Extension DN, it will attempt to route the call to another DN (or default-dn) on expiration of this timer. You must set this timer to be less than the parameter rq-expire-tmout value of 32000 (32 seconds).
  • SIP Server is able to start properly with the proper FlexLM license installed.

4. Genesys Media Server Deployment

Follow these steps to configure a Media Server deployment.

Media Server platform consists of Resource Manager and Media Control Platform applications in the Genesys Voice Platform product suite. To deploy, Media Control Platform and optionally a Resource Manager are required to be installed. When installed, Resource Manager serves as the ingress point to Media services and provides a MCP resource as a media service to the network/calling side.

  1. Install and configure MCP (Media control Platform) using the Genesys Media Server Deployment Guide.
  2. Within the MCP application’s Connections tab, add connections to SNMP Master Agent, Message Server, and Reporting Server (optional).

    The connections to applications are added for the following reasons:

    Message Server - To ensure that component log information reaches the Log database and can be viewed in the Solution Control Interface (SCI)

    Reporting Server - (Optional) To ensure that these components detect the Reporting Server to which they are sending reporting data.

    SNMP Master Agent - To ensure that alarm and trap information is captured.

  3. Verify VoIP service DN of type=msml as specified in section Creating SIP Switch in Genesys Administrator to support SIP Server-Media Server MSML interactions to support treatments and conferencing capabilities.
  4. To play music on hold (MOH) and music treatments, verify the following options are set in MCP and SIP Server:

    MCP->msml-> play.basepath = file://$InstallationRoot$ (this is the installation folder of Media Server. After this is, it will automatically look for the music sub folder).

    “MOH” and music treatments are located in the “music” folder.

    The ‘announcement” folder should contain ‘prompt’ files with proper IDs to support. Used in the URS Routing Strategies as mentioned in chapter 0.

    SIP Server->TServer->msml-support=true

  5. Install and configure Resource Manager as per Genesys Media Server Deployment Guide.

    Note: If SIP Server and Resource Manager are on the same machine and within the Resource Manager application, then the default SIP listening port number should be increased by 100 so the Resource Manager listening port is set to 5160 and the SIP Server application listens on port 5060. Make the necessary port changes within Resource Manager’s sip, proxy, register, subscription, and monitor sections.

  6. Within the Resource Manager application’s Connections tab, add connections to SNMP Master Agent, Message Server, and Reporting Server (optional).

    The connections to applications are added for the following reasons:

    Message Server - To ensure that component log information reaches the Log database and can be viewed in the Solution Control Interface (SCI).

    Reporting Server - To ensure that these components detect the Reporting Server to which they are sending reporting data. (Optional).

    SNMP Master Agent - To ensure that alarm and trap information is captured.

  7. Within the Integrating Media Control Platform with the Resource Manager, click the Media Control Platform Application object. TheConfiguration tab appears.

    Click the Options tab, and use the View drop-down list to select Show options in groups...

    Select sip to find the routeset option.

    In the Value field, type the following:

    <sip:IP_RM:SIPPort_RM;lr>

     

    Where IP_RM is the IP address of the Resource Manager, and SIPPort_RM is the SIP port of the Resource Manager—typically, 5060.

     

    Note: You must include the angle brackets in the Value field in the sip.routeset and sip.securerouteset parameters.

     

    In the Value field of the securerouteset option, type the following:

    <sip:IP_RM:SIPSecurePort_RM;lr>

  8. G.729 media codec is not configured by default as a supported codec or as a codec that can be transcoded. This support can be enabled by adding “g729” as one of the values to the mpc.codec and mpc.transcoders space separated list.  The G.729 media codec was not provisioned in this Genesys deployment and is only mentioned here for completeness.

    Example:

    mpc.transcoders=PCM GSM G726 G729

    mpc.codec=g729 pcmu pcma g726 gsm h263 h263-1998 h264 telephone-event

     

    Alternately Media Server (specifically MCP component) can be configured to respond a multiple codec offer request with a single codec response. This feature support is available starting with MCP 8.1.4 release.

    This setting can be enabled by setting mpc.answerwithonecodec=1 (Default=0 – MCP responds to multiple codec offer with a multiple codec response list).

  9. This step is optional and is only required if multiple media control platform (MCP) instances are deployed and need to be controlled by Resource Manager for load balancing.

 

Log in to Genesys Administrator.

  1. On the Provisioning tab, click Voice Platform>Resource Groups.

  2. On the Details pane tool bar, click New.

  3. The Resource Group Wizard opens to the Welcome page.

  4. On the Resource Manager Selection page, add the Resource Manager Application object for which you want to create the group. On the Group Name and Type page: enter MCPGroup or any custom name without spaces. Select type as Media Control Platform.

  5. On the Tenant Assignments page, add the child tenant to which the Resource Group will be assigned.

    • Note: -The above bullet item is required only if you are creating the Resource Group in a multi-tenant environment.

  6. On the Group Properties page, enter the information as specified below for the Resource Group that you are configuring.

  • Monitoring Method: Retain the default value: SIP OPTIONS.

  • Load Balance Scheme: Select round-robin.

  • Port Usage Type: Select in-and-out.

  • Maximum Conference Size: Enter -1.

  • Maximum Conference Count: Leave blank.

 

Note: For the Media Control Platform group, the Max.Conference Size and Max.Conference Count, and the Geo-location options are optional.

For a complete list of resource-group options and their descriptions, refer to  the Genesys Voice Platform User’s Guide.

In this step, you create a default IVR Profile that can be used to accept calls other than those specified in the dialing plans.

  1. Log in to Genesys Administrator.
  2. On the Provisioning tab, select Voice Platform>IVR Profiles.
  3. In the Tasks panel, click Define New IVR Profile. The IVR Profile Wizard opens to the Welcome page.
  4. On the Service Type page, enter the name of the default IVR Profile, IVR_App_Default.
  5. Select either Conference or Announcement from the drop-down list. (Only one service type per IVR Profile is supported.)
  6. If you selected Conference, on the Service Properties page, enter a conference ID number.
  7. If you selected Announcement, on the Service Properties page, enter the URL of the announcement, for example, http://webserver/hello.wav.
  8. Click Finish.
    • Note: When you use the IVR Profile Wizard to create the default profile, the gvp.general and gvp.service-prerequisites sections are created for you and include the required parameters
  9. In the gvp.general section of the Tenant’s Annex tab, set the default-application to this default IVR Profile name – IVR_App_Default.

This completes installation and configuration of Media Server. Make sure Resource Manager and MCP are started successfully.

5. Stat Server Configuration

This section explains configuration of Stat Server that connects with T-Servers/SIP Servers and maintains agent and/or extension status which is used by URS during call routing.

  1. Install and configure Stat Server per the Genesys Framework Stat Server Deployment Guide.
  2. Add connections to SIP Server, Message Server to perform real-time monitoring of the SIP agent status.

Figure : Genesys Administrator - Stat Server Connections

6. Universal Routing Configuration in Genesys Administrator

This section explains how to configure a Universal Routing Configuration (URS) to support execution of call routing on SIP Server.

  1. Install and configure Universal Routing Server per the Genesys Universal Routing Deployment Guide.
  2. Add connections to Message Server, Stat Server, and SIP Server.
  3. Add connection to SIP Server to monitor events received by SIP Server for various route points and extensions on the SIP Server Switch.
  4. Add connection to Stat Server to query Stat Server for routing calls to available and ready agents.

Figure : Genesys Administrator - URS Configuration


Use any of the strategies below to test your configuration. 

7. URS Routing Strategies

This section shows examples of five URS routing strategies used during testing. 

  • Strategy #1 - Route Call to Available Agent

    When RP 1011 is invoked, this strategy routes the call to the next available agent. If no agent is available, it plays MOH until one becomes available.

Figure : URS Strategy #1


 

  • Strategy #2 - Play Announcement and Route to Available Agent

    When RP 1012 is invoked, this strategy plays an announcement and routes the call to the next available agent. If no agent is available, it plays MOH until an agent becomes available.

Figure : URS Strategy #2


Figure : URS Strategy #2 - Target Selection


 

  • Strategy #3 – Play Announcement and Collect Seven Digits

    When RP 1013 is invoked, this strategy verifies that any seven digits can be collected and then routed to an available agent. If no agent is available, it plays MOH until an agent becomes available.

Figure : URS Strategy #3

Figure : URS Strategy #3 - Prompt Tab

Figure : URS Strategy #3 - Busy Tab

Figure : URS Strategy #3 - Target Selection

  • Strategy #4 – Route to External SIP Carrier Number

    When RP 1014 is invoked, this strategy immediately routes the call to an external SIP Carrier number.

Figure : URS Strategy #4

 

  • Strategy #5 – Route to External SIP Carrier Number

    When RP 1015 is invoked, this strategy plays an announcement and then immediately routes the call to an external SIP Carrier number.

Figure : URS Strategy #5

Figure : URS Strategy #5 - Prompt Tab

Figure : URS Strategy #5 - Function Properties

 

 

Sonus SBC 5xx0 Series Configuration

 

Complete Configuration
configure
#DSP Resources
set system mediaProfile compression 90 tone 10
commit
#Element Routing Priority profile
set profiles callRouting elementRoutingPriority TG_ERP entry localOperator 0 entityType trunkGroup
set profiles callRouting elementRoutingPriority TG_ERP entry nationalType 0 entityType trunkGroup
set profiles callRouting elementRoutingPriority TG_ERP entry internationalType 0 entityType trunkGroup
commit
#Ip Signaling Profiles
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes flags endToEndBye enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags dialogEventPackage enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags info enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags refer enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags statusCode3xx enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile GENESYS_IPSP egressIpAttributes flags disable2806Compliance enable
commit
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags disableMediaLockDown enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags reQueryPsxOnRegisterRefresh enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags endToEndBye enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags relayDataPathModeChangeFromOtherCallLeg enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags dialogEventPackage enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags info enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags options enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags refer enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags statusCode3xx enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags updateWithoutSdp enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags authcodeHeaders enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes sipHeadersAndParameters flags endToEndAck enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes flags sameCallIdForRequiredAuthorization enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes privacy transparency enable
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags sendSdpInSubsequent18x enable
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags suppress183WithoutSdp enable
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags suppress183For3xxRedirectResponse enable sendSdpIn200OkIf18xReliable enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags noPortNumber5060 enable
commit
#Trusted Side Configuration
#IP Interface Group
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED portName pkt0
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED ipAddress 10.35.177.246
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED prefix 26
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED state enabled
commit
#IP Static Route
set addressContext default staticRoute 0.0.0.0 0 10.35.177.193 TRUSTED TRUSTED preference 100
commit
#zone
set addressContext default zone TRUSTED id 3
commit
#SIP signaling port
set addressContext default zone TRUSTED sipSigPort 3 ipInterfaceGroupName TRUSTED
set addressContext default zone TRUSTED sipSigPort 3 ipAddressV4 10.35.177.247
set addressContext default zone TRUSTED sipSigPort 3 portNumber 5060
set addressContext default zone TRUSTED sipSigPort 3 transportProtocolsAllowed sip-udp
set addressContext default zone TRUSTED sipSigPort 3 mode inService
set addressContext default zone TRUSTED sipSigPort 3 state enabled
commit
#IP Peer
set addressContext default zone TRUSTED ipPeer GENESYS ipAddress 10.35.176.111
set addressContext default zone TRUSTED ipPeer GENESYS ipPort 5060
set addressContext default zone TRUSTED ipPeer GENESYS authentication intChallengeResponse enabled
set addressContext default zone TRUSTED ipPeer GENESYS authentication incInternalCredentials enabled
commit
#SIP trunk group
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS policy callRouting elementRoutingPriority TG_ERP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS policy signaling ipSignalingProfile GENESYS_IPSP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS media mediaIpInterfaceGroupName TRUSTED
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS ingressIpPrefix 10.35.176.111 32
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS mode inService
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS state enabled
commit
#Untrusted Side Configuration
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED ceName LITTLE
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED portName pkt2
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED ipAddress 10.35.177.150
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED prefix 26
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED mode inService
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED state enabled
commit
#IP Static Route
set addressContext default staticRoute 0.0.0.0 0 10.35.177.129 UNTRUSTED UNTRUSTED preference 100
commit
#zone
set addressContext default zone UNTRUSTED id 4
#SIP signaling port
set addressContext default zone UNTRUSTED sipSigPort 4 ipInterfaceGroupName UNTRUSTED
set addressContext default zone UNTRUSTED sipSigPort 4 ipAddressV4 10.35.177.151
set addressContext default zone UNTRUSTED sipSigPort 4 portNumber 5060
set addressContext default zone UNTRUSTED sipSigPort 4 transportProtocolsAllowed sip-udp,sip-tcp
set addressContext default zone UNTRUSTED sipSigPort 4 mode inService
set addressContext default zone UNTRUSTED sipSigPort 4 state enabled
commit
#IP Peer
set addressContext default zone TRUSTED ipPeer PSTN ipAddress 10.35.8.146
set addressContext default zone TRUSTED ipPeer PSTN ipPort 5060
set addressContext default zone TRUSTED ipPeer PSTN surrogateRegistration authUserName 2086041020
set addressContext default zone TRUSTED ipPeer PSTN surrogateRegistration regAuthPassword 123456
set addressContext default zone TRUSTED ipPeer PSTN surrogateRegistration sendCredentials challengeForAnyMessage
set addressContext default zone TRUSTED ipPeer PSTN authentication intChallengeResponse enabled
set addressContext default zone TRUSTED ipPeer PSTN authentication incInternalCredentials enabled
commit
#SIP trunk groups
#PSTN trunk
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN policy callRouting elementRoutingPriority TG_ERP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN policy signaling ipSignalingProfile GENESYS_IPSP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN media mediaIpInterfaceGroupName UNTRUSTED
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN ingressIpPrefix 10.35.8.146 32
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN mode inService
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN state enabled
commit
#RDN trunk
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN policy callRouting elementRoutingPriority TG_ERP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN policy signaling ipSignalingProfile ACCESS_IPSP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN signaling registration requireRegistration required
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN media mediaIpInterfaceGroupName UNTRUSTED
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN ingressIpPrefix 0.0.0.0 0
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN mode inService
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN state enabled
commit
#Global Configuration
#Route Labels
set global callRouting routingLabel TO_TWO_WAY_GENESYS routingLabelRoute 0 routeType trunkGroup
set global callRouting routingLabel TO_TWO_WAY_GENESYS routingLabelRoute 0 trunkGroup TWO-WAY-SIP-GENESYS
set global callRouting routingLabel TO_TWO_WAY_GENESYS routingLabelRoute 0 ipPeer GENESYS
commit
set global callRouting routingLabel TO_TWO_WAY_PSTN routingLabelRoute 0 routeType trunkGroup
set global callRouting routingLabel TO_TWO_WAY_PSTN routingLabelRoute 0 trunkGroup TWO-WAY-SIP-PSTN
set global callRouting routingLabel TO_TWO_WAY_PSTN routingLabelRoute 0 ipPeer PSTN
commit
#Routes
set global callRouting route trunkGroup TWO-WAY-SIP-GENESYS LITTLE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_PSTN
set global callRouting route trunkGroup TWO-WAY-SIP-GENESYS-RDN LITTLE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_GENESYS
set global callRouting route trunkGroup TWO-WAY-SIP-PSTN LITTLE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_GENESYS
commit
exit

 

Test Results

Table : Test Results

S.NoProcedureObservationResultComment
Inbound calls     
1Inbound Call to Agent released by caller Pass 
2Inbound Call to Agent released by agent Pass 
3Inbound Calls rejected Pass 
4Inbound Call abandoned Pass 
5Inbound Call to Route Point with Treatment Pass 
6Interruptible Treatment Pass 
7IVR (Collect Digit) Treatment Pass 
8Inbound Call routed by using 302 out of SIP Server signaling path Pass 
Outbound Calls    
91PCC Outbound Call from SIP Endpoint to external destination Pass 
103PCC Outbound Call to external destination Pass 
111PCC Outbound Call Abandoned Pass 
Music on hold    
12Caller is put on hold and retrieved by using RFC 2543 method Pass 
13T-Lib-Initiated Hold/Retrieve Call with MOH using RFC 3264 method Pass 
Call Transfers    
143PCC 2 Step Transfer to internal destination by using re-INVITE method Pass 
153PCC Alternate from consult call to main call Pass 
161PCC Unattended (Blind) transfer using REFER Pass 
171PCC Attended Transfer to external destination Pass 
183PCC Two Step Conference to external party Pass 
193PCC (same as 1PCC) Single-Step Transfer to another agent Pass 
203PCC Single Step Transfer to external destination using REFER Pass 
213PCC Single Step Transfer to internal busy destination using REFER Pass 
Other scenarios    
22Early Media for Inbound Call to Route Point with Treatment Pass 
23Early Media for Inbound Call with Early Media for Routed to Agent Pass 
24Inbound call routed outbound (Remote Agent) using INVITE without SDP Pass 
25Call Progress Detection Pass 
26Out of Service detection. Checking MGW live status Pass 
27SIP Authentication for outbound calls Not Supported 
28SIP Authentication for incoming calls Pass 
Remote Agent behind SBC scenarios    
29T-Lib-Initiated Answer/Hold/Retrieve Call for Remote SIP endpoint which supports the BroadSoft SIP Extension Event Package Pass 
303PCC Outbound Call from Remote SIP endpoint to external destination Pass 
313PCC 2 Step Transfer from Remote SIP endpoint to internal destination Pass 
321PCC Attended Transfer from Remote SIP endpoint to external destination Pass 
 

Conclusion

These Application Notes describe the configuration steps required for Sonus SBC 5XX0 to successfully interoperate with Genesys Voice Platform. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.

 

 

Appendix A

The following new configurations are included in this section:

  1. SIP Server standard configuration
  2. DN Configuration
  3. EpiPhone configuration

1. SIP Server standard configuration

Information below displays how objects are configured by default.

Table : SIP Server standard configuration

SIP Server Application CME Options TServer section

SIP-hold-rfc3264=true

router-timeout=30

default-dn=

blind-transfer-enabled=true

resource-management-by-rm=true

msml-support=true

sip-enable-moh=true


2. DN Configuration 

 

Table : DN Configuration

Name

Number

Name in CME

CME Options TServer section

Comment

MGW-TRUNK

MGW-TRUNK

MGW-TRUNK

refer-enabled=true

contact=<TSE_CONTACT>

oos-check=10

oos-force=5

oosp-transfer-enabled=true

sip-replaces-mode=2

TSE

Ext-DN1

Ext-DN2

21001

21002

N/A

N/A

 

SIP-DN1

SIP-DN2

7101

7102

7101

7102

refer-enabled=false

ring-tone-on-make-call=false

make-call-rfc3725-flow=1

contact=*

 

SIP-RDN

7200

7200

refer-enabled=true

ring-tone-on-make-call=false

make-call-rfc3725-flow=1

contact=*

sip-cti-control=talk,hold

SIP endpoint which supports the BroadSoft SIP Extension Event Package.

SIP-UNKN

7777

N/A

N/A

 

RP

5000

5000

 

 

RP1

5001

5001

 

 

RP2

5002

5002

 

 

SVC_MSML

SVC_MSML

SVC_MSML

prefix=msml=

contact=<MS_CONTACT>

service-type=msml

subscription-id= Environment

MS

3. EpiPhone configuration


Content of configuration file esttt.conf:

Table : EpiPhone configuration

 [TcCM]

site1 = UTE_HOME

connect-on-startup  = true

open-log-on-startup = false

log-to-file = epi-phone.log

#------------------------------------------------------------------------------

[UTE_HOME]

server = (host=<SIP_SERVER_HOST_IP>,port=<SIP_SERVR_TLIB_PORT>)

sip-register = false

dn1 = 7101,name="Alice",mkcall="7102"

dn2 = 7102,name="Bob"

dn3 = 7200,name="John"

dn4 = 5000,pool="shared"

dn5=5001,pool="shared",script="annc=(PROMPT=(\"1\"=(INTERRUPTABLE=1,ID=1)))"

dn6=5002,pool="shared",script="collect=(MAX_DIGITS=4,RESET_DIGITS=11,BACKSPACE_DIGITS=22,TOTAL_TIMEOUT=100)"