Recorder Integration Service

Genesys Integration Guide

Version 15.2
Document Revision 1.11
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Document Revision 1.11
Published November 05, 2019
Customer Engagement SolutionsTM
 C o n t e n t s

About this guide 6

Genesys integration with Recorders 8
Genesys product overview 9
Genesys SIP Server 9
Genesys Platform SDK 9
GVP Media Server 9
GVP Resource Manager 9
Genesys Interaction 10
Full-time vs. Selective Recording 10
Full-time Recording 10
Selective Recording 10
Third-party components for VoIP Gateway Recording 11
Supported integrations 12
Supported recording solutions 13
Recorder Platform Components 14
Genesys SIP Server Integration 15
Recording Solution 16
VoIP Station-side Interception 16
VoIP Station-side Delivery 16
VoIP Gateway Recording 17
SIPREC 17
Session Replication with Oracle SBC 19
Requirements 20
Limitations/Known Issues 21
Unable to replay session of second agent in three-way conference with Oracle SBC 21
Recording of Internal and Consultation Calls in VoIP Delivery 21
Internal or consultation call between monitored agents 21
Recording of Internal and Consultation Calls in VoIP Delivery 21
Internal or consultation call between monitored agent and non-monitored agent 21
SIP Trunk Recording with Oracle SBC 21
SIPREC 22
Genesys Recording Indication Feature 22
Configuration Checklist 23
Licensing 26
Configuration 27
 Switch Configuration 27
VoIP Station-side Interception 27
VoIP Station-side Delivery 27
Configure Oracle SBC for SRR 29
Configure AudioCodes or Oracle SBC for SIPREC 29
Secure RTP with SIP over TLS 29
Recorder Configuration 30
VoIP Station-side Interception 30
VoIP Station-side Delivery 32
VoIP Gateway Recording 32
Certificates for Genesys Platform SDK and Interaction Server 36
CTI Adapters 38
Create a Genesys Voice Platform SDK Adapter 39
Create a Genesys (Legacy T-lib) Adapter 41
SIP Proxy 43
SIPREC Adapter 45
Advanced Deployments 48
Genesys IVR Recording 48
Call Recording Flow 48
Configure the Recorder for IVR 49
Configure contact grouping 50
Maintenance 51
Troubleshooting 52
Failed to Connect to Genesys T-Server 52
Troubleshooting 52
Failed to Connect to Genesys T-Server 52
Incorrect Call Tagging 53
Call Type (Direction) tagged as "Unknown" 53
Incorrect Call Tagging 53
Call Type (Direction) tagged incorrectly 53
Wrong extension number tagged; calls not recorded 53
Failed to Record Calls in IP Interception Environment 54
Failed to Record both parties on the call in Genesys Delivery 54
Genesys Business Continuity Integration 55
Genesys Business Continuity Overview 56
Fallback behavior 57
Configure Genesys Business Continuity 58
Genesys Chat Integration 59
Genesys Chat Integration Overview 60
Security 60
Redundancy 60

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 Workflow: Configure the integration 61
Obtain Genesys Interaction Server information 61
Configure Recorder roles 62
Create a Genesys Chat data source 62
Add agents to the system 63
Genesys Chat 64
High Availability for Voice and Screen 67
N+N Redundancy 68
N+M Redundancy 69
N+M All Shared Scenario 69
1+1 Integration Service Redundancy with VoIP Delivery 70
N Dedicated + M Shared Scenario 70
Real-time/Record-on-Demand 71
Using Corero and/or Link Protectors in an N+M All Shared Environment 72
SIP Recording Configuration 72
Oracle SBC Redundancy 74
Oracle SBC Load Balancing 75
Genesys High Availability and Disaster Recovery 76
Genesys T-Server Hot and Warm Standby 76
SIP Business Continuity Deployment 76
VoIP Station-side Interception 77
VoIP Station-side Delivery (SIP Trunk with Oracle SBC or SIP Trunk Interception) 78
High Availability for Text 79
Overview 80
Configure High Availability for Text 81
Set Up Certificates 82
Overview 83
Certificates for Genesys Platform SDK and Interaction Server 84
Certificates 84
Authentication 84
Workflow 85
Certificates for Genesys Platform SDK and Interaction Server 86
Certificates 86
Authentication 86
Workflow 87
Provide a certificate to Genesys 88
Configure the adapter for TLS 89
Certificates for SBC 90
Set up keys for certificates 90
Use existing keys 93
Provide your own keys 93

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 P r e f a c e

About this guide

This guide documents Recorder integration with Genesys, including specific ways in which
configuration and behavior differs from that documented in the Recorder Configuration and
Administration Guide, to which you should also refer.

Intended Audience
This guide is designed for:
• Verint Field Services and Support personnel.
• Customers responsible for site preparation and planning.

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To learn more about the support options that best suit your needs, visit us at Customer
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About this guide

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To learn more about available training options from Verint Systems, visit us at verint.com/training.

Revision Description of changes

1.11 Revised Genesys Interaction description.

1.10 Added support for route point tracking.

1.09 Configuration for Genesys Business Continuity no longer requires that you
configure the same set of extensions twice.

1.08 Removed duplicate topics.

1.07 Corrected placement of SIPREC and SIP Proxy adapter topics.

1.06 Added support for Genesys Chat.

(HFR5)

1.05 Added support for Genesys Business Continuity.

(HFR4)

1.04 Updated with new document template.

(HFR3)

1.03 Revised guidance on configuring UUIDs in custom attributes, associating data

sources with a primary adapter, and port calculation.

1.02 Removed AutoDetectProxies, PersistDetectedProxies, and NeverExpires SIP

settings, as they are no longer applicable.

1.01 Added support for TLS authentication with Genesys.

1.00 Initial version. New in this release:

Support for Oracle SBC and TLS/SRTP support with SIPREC.

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 C h a p t e r 1  

Genesys integration with

Recorders

This chapter describes supported recording solutions and the components involved in these
integrations.

Topics 

Genesys product overview 9

Third-party components for VoIP Gateway Recording 11
Supported integrations 12
Supported recording solutions 13
Recorder Platform Components 14
Genesys integration with Recorders Genesys product overview

Genesys product overview

This guide covers integrations with the Genesys Session Initiation Protocol (SIP) Server via the Genesys
Platform SDK.

Genesys SIP Server

The Genesys SIP Server is a combined T-Server and call switching component that provides an interface
between telephony hardware and other Genesys software components in the enterprise. In giving the
entire line of Genesys products access to SIP networks, the SIP Server offers a standards-based, platform-
independent means of leveraging voice/data convergence.
The SIP Server:
• provides all SIP signaling and T-Server functions.
• tracks events and requests to and from the telephony device as contacts flow through your
enterprise.
• acts as a messaging interface between SIP Server clients.
• performs call switching and control, meaning no third-party PBX or ACD system is required.

Genesys Platform SDK

The Genesys Platform SDK is a client library deployed in order to communicate with the Genesys SIP
Server. It provides the ability to:
• Monitor and control extensions involved in call center transactions.
• Receive call metadata from the SIP Server.

GVP Media Server

In VoIP Delivery, the Genesys Voice Platform (GVP) Media Server performs the actual file-based recording
or replicates the media stream to the Recorder by establishing the recording session to the recording
server function. The SIP Server initiates call recording on the GVP, using media control to direct media
towards the Media Server. The Media Server then negotiates the media between the endpoints in order
to minimize the need for transcoding and preserve security of the audio stream.

GVP Resource Manager

In VoIP Delivery, the GVP Resource Manager manages a pool of media servers within GVP and applies
runtime policies, such as ensuring call legs to the same conference are pinned to the same media server. It
is also the SIP proxy for the recording session, so that the Resource Manager hides the high availability
details of each third-party vendor from other Genesys components such as the Media Server.

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Genesys integration with Recorders Genesys product overview

Genesys Interaction
The Genesys Interaction recording solution allows organizations to capture complete interactions from
any channel, including communications across chat.

Full-time vs. Selective Recording

The Genesys Call Recording solution supports both Full-time and Selective Recording.

Full-time Recording
In Full-time VoIP Recording, Genesys is configured to automatically initiate (via a SIP INVITE) recording on
a per call basis for all monitored extensions.

Selective Recording
In Selective VoIP Recording, the Integration Service initiates media streaming and Selective Recording is
performed by allocating recording resources through the SDK adapter (which receives requests from the
Genesys Voice Platform SDK), per recorded call.
The Genesys SIP Server can be configured to select a Media Server to transmit audio for delivery, based
on extension. (It is not possible to select a Media Server based on call origin.)
To use the recording resources closest to the Media Server for recording, you must create an Extension
Pool Member Group for each Media Server. The Media Server must be configured with its assigned
Recorders, and the list of extensions Genesys has associated to it. This ensures that the system selects
the correct recording resources at the same location that Genesys has configured the recording to
originate from.
Each IP Recorder is configured with a range of ports that serve as destinations for extensions’ audio,
which is duplicated and delivered by the Genesys T-Server to the IP Recorder. The Integration Service
allocates these ports for recording, and maintains a list of available ports and used ports.
 

 With Selective Recording, the number of recording resources (IP Recorder ports) should not be
less than the number of target extensions that need to be recorded simultaneously. However,
the number of recording resources can be less than the number of all the extensions in the
contact center.

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Genesys integration with Recorders Third-party components for VoIP Gateway Recording

Third-party components for VoIP Gateway

Recording
You may use the Genesys SIP with an Oracle (formerly Acme Packet) Session Border Controller (SBC) (for
either session replication or Session Initiation Protocol Recording [SIPREC]) or an AudioCodes Gateway
(for SIPREC). See VoIP Gateway Recording, page 17 for more details.

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Genesys integration with Recorders Supported integrations

Supported integrations
The table below lists the Genesys/CTI Server combinations and the recording solutions
supported when integrating with Verint Recorders.
A

Switch/CTI Trunk- Station/ CTI- Recorder- VoIP SIP Trunk

Side Extension- controlled controlled Delivery Recording
TDM Side TDM IP* IP

Genesys SIP     P 
Server

* CTI Controlled Recording is supported in environments where Genesys can provide IP information for
the endpoints in the call. Be sure that you have configured the environment to connect to the SIP Server
against which the phones are registered.

Multiple adapters
You can have more than one adapter in your enterprise, and you can associate multiple adapters with a
single data source. However, a single data source should be assigned only one "primary" CTI adapter. This
CTI adapter is defined as the Primary controlling CTI adapter (not tagging only). SIP proxy adapters are not
considered CTI adapters.

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Genesys integration with Recorders Supported recording solutions

Supported recording solutions

The following table describes recording solutions supported for Genesys Integrations. For further
information on recording solutions refer to the relevant Recording Deployment Reference Guides.

Recording Description and documentation

solution

VoIP Interception WFO intercepts (monitors or “sniffs”) the contact center's VoIP traffic and
records the IP packets of interesting calls, based on rules.

VoIP Delivery In a VoIP contact center, the IP traffic is monitored and recorded. When
recording is triggered, the Recorder assembles the relevant packets and
stores the contacts in standard audio file formats. This integration
environment offers Full and Selective VoIP Delivery. See the Recorder VoIP
Delivery Deployment Reference Guide.

VoIP Gateway SRR with Oracle SBC

Recording
SRR with Oracle SBC uses RFC 2003 for Interception. RFC 2003 is a method
of en of an IP datagram within another IP datagram.

SIP Trunk Delivery

The Recorder records traffic at the SIP Trunk. This includes SIPREC
deployments, or environments in which SIP trunk sessions are replicated by
an edge device such as Oracle SBC to the Recorder.

SIP Trunk Interception

The Recorder intercepts (monitors or “sniffs”) the contact center's SIP traffic
at the SIP Trunk and records the packets of interesting calls, based on rules.

   

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Genesys integration with Recorders Recorder Platform Components

Recorder Platform Components

The following server roles are required to support this environment:
• Recorder Integration Service - acts as the interface between the recording system and output
from the switch and other data sources. The Integration Service processes events from the server
interfaces, detects state or data changes, and passes them along to other subsystems. It can
capture CTI event streams to file for later playback and viewing, and holds agent state, device
state, call state, and data associated with all known devices and calls.
• IP Recorder - used for recording audio from IP data sources, such as switches and IP gateways.
The system supports two types of IP recording: Gateway and Extension-Side.
• Screen Recorder - used to record screen activity.
See the Enterprise Manager Configuration and Administration Guide for more information on server roles.

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 C h a p t e r 2  

Genesys SIP Server Integration

This chapter describes how to install and configure a Genesys SIP Server integration via the
Genesys Platform SDK.

Topics 

Recording Solution 16
Requirements 20
Limitations/Known Issues 21
Configuration Checklist 23
Licensing 26
Configuration 27
Advanced Deployments 48
Maintenance 51
Troubleshooting 52
Genesys SIP Server Integration Recording Solution

Recording Solution
This integration supports VoIP Station-side Interception, VoIP Station-side Delivery, and VoIP Gateway
Recording.

VoIP Station-side Interception

VoIP Interception is VoIP recording using a SPAN port/port mirror. With VoIP Interception, spanning can
be accomplished from one of three general locations within a customers network topology. These
locations are at the gateway, switch side (infrastructure), and station side. With gateway recording the
VoIP traffic is spanned at a gateway typically located at the point where the call enters the customer’s
environment. Switch-side (infrastructure) recording is accomplished by spanning the traffic on the station
side of the switch. Station-side recording is accomplished by spanning the data switch that the agent
phones are on.

 Please see the Recorder VoIP Interception Deployment Reference Guide for more information.

VoIP Station-side Delivery

VoIP Delivery is supported in SIP-driven DMS for Genesys. The Recorder Integration Service serves as a
SIP end point to negotiate the audio delivery for all monitored extensions. The following diagram
illustrates the Recorder integration with the Genesys Call Recording solution and describes the flow of
recording initiation.

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Genesys SIP Server Integration Recording Solution

1 Party A establishes a call with Party B.

Both parties communicate with the SIP Server using SIP.
2 The SIP Server reports a call to the Integration Service
• In Full and Selective Delivery, the SIP Server reports a call to the Integration Service via the
Genesys Voice Platform SDK or T-Lib software.
• In Selective Delivery, the Integration Service returns a reply to SIP Server.
The SIP Server directs the media of Party A and Party B towards the Media Server.
The SIP Server uses media control to instruct the Media Server to replicate media stream towards
the Media Recorder.
3 The Media Server receives the media streams and establishes a recording session (a SIP dialog) with
the Integration Service, in order to negotiate audio delivery of a call. The Integration Service provides
the IP address of the IP Recorder and the ports for audio delivery.
4 The Media Server delivers the RTP of party A and party B to the IP Recorder.
 

VoIP Gateway Recording

There are three types of VoIP Gateway Recording.
• SIPREC—SIPREC is a recording standard and a type of VoIP Delivery recording. SIPREC traffic may
come from a switch, or from a device like the AudioCodes Gateway or Oracle SBC. These
components act as a Session Recording Client (SRC) between the Recorder Integration Service and
the SIP Trunks/SIP phones/call center system.
• SIP Session Replication for Recording—In SIP Session Replication for Recording (SRR), the
Oracle SBC encapsulates and directs both the SIP and RTP to a particular IP Recorder. If the
Recorder fails to respond correctly, the SBC can route the SIP and RTP to a secondary Recorder in
the configuration.
• SIP Trunk Delivery via Genesys SIP Server—Genesys Delivery forks calls from a trunk DN of
the phone DN (no SBC is involved). Correlation to CTI is performed in a manner similar to that
used for SIPREC.

SIPREC
Session Initiation Protocol Recording (SIPREC) is a recording standard and a type of VoIP Delivery
recording. The integration may receive SIPREC traffic from a switch, or session replicating device like the
AudioCodes Gateway or SBC. This component acts as the Session Recording Client (SRC) between the
Recorder Integration Service and the SIP Trunks/SIP phones/call center system.
In SIPREC, an edge device routes SIP signaling to a particular “recorder” (technically the SIPREC adapter in
the Recorder Integration Service). The Recorder Integration Service receives and manages the SIP calls,
routing the RTP to an associated IP Recorder, and tagging appropriate information extracted from the SIP
signaling. Default tagging includes any custom attributes, any relevant data found in the SIPREC XML, as
well as the data source ID and member group ID. The Recorder Integration Service uses this tagged
information to correlate the call with the CTI.
Both full-time and selective recording are supported in SIPREC.
• Full-time—Records all calls for which SIP is delivered to the Recorder Integration Service.

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Genesys SIP Server Integration Recording Solution

• Selective—Selective SIP Recording is ideal for environments where not all calls need to be recorded
(including otherwise full-time environments with a large number of Interactive Voice Response
(IVR) ports that don't need to be recorded). Recording Rules and commands received via third-
party APIs can be used to refine which particular calls are recorded, resulting in reduced traffic
volume/resource use.
In most Genesys environments, there are multiple servers handling the , and in these cases the Genesys
CallUUID does not remain constant. Each Genesys server that handles the call assigns its own CallUUID. As
the call is transferred between servers, new UUIDs are assigned to the call. However, since the transfer is
accomplished only by negotiation between Genesys servers, Genesys does not “re-INVITE” the SBC (thus
the SBC is not notified of the new UUID values). The end result is that by the time the call reaches the
agent, the CTI events standard CallUUID field only contains the last UUID assigned by the Agent SIP server
with which the agent is registered. The original CallUUID is no longer present in the CTI events.
To resolve this requires the creation of a new “custom CallUUID” in both SIP and CTI. To accomplish this
you may use the following process:

On the INBOUND CALL

1 Using Oracle SBC Header Manipulation Rules, SBC acquires initial X-Genesys-CallUUID on OK response
to initial INVITE to Genesys GVP.
2 Using Oracle SBC Header Manipulation Rules, SBC copies initial X-Genesys-CallUUID into new SIP
header field (let’s call this “Verint-UUID” for this example).
3 SBC includes “VerintUUID” in the SIPREC INVITE to Verint as SIP Header or SIPREC XML Metadata.
4 Genesys acquires custom SIP header “VerintUUID” and attaches it to the call metadata as Genesys
UserData key/value pair (“VerintUUID”).
5 As call is transferred to Agent SIP Server, UserData is carried and comes to Verint via SDK CTI Events
(Established, etc.) as UserData key/value field.

On OUTBOUND CALL
1 Using Oracle SBC Header Manipulation Rules, SBC acquires initial X-Genesys-CallUUID on initial INVITE
from Genesys GVP.
2 Using Oracle SBC Header Manipulation Rules, SBC copies initial X-Genesys-CallUUID into new SIP
header field (“Verint-UUID”).
3 SBC includes “VerintUUID” in SIPREC INVITE to Verint as SIP Header or SIPREC XML Metadata.
4 Genesys acquires custom SIP header “VerintUUID” and attaches it to the call metadata as Genesys
UserData key/value pair (“VerintUUID”).
5 UserData is carried and comes to Verint via SDK CTI Events (Established, etc.) as UserData key/value
field.
Once this is complete, you can then configure separate custom attributes for the SIP VerintUUID and CTI
VerintUUID used in the Correlation Key Configuration in the Gateway Side Correlation Member Group
settings.

Related information 
"SIP Trunk Delivery" in the Recorder VoIP Delivery Deployment Reference Guide.

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Genesys SIP Server Integration Recording Solution

Session Replication with Oracle SBC

SRR via an Oracle SBC allows a copy of specific calls to be delivered to a recording device on the network.
SBC-based recording is common in Avaya/SIP environments, and can simplify the administration and
management of SIP trunk recording.
The Recorder supports integration with the Oracle standalone SBC 3810/3820 and 4250/4500 product
line in VoIP Interception environments. The replicated calls are encapsulated using RFC 2003 and
delivered to the Recorder, where the traffic is decoded and recorded.
 

 This release does not support integration with the 2600-series SBC.

The benefits of integrating your Recorder with SBC include:

• the ability to configure SBC to fork only a defined subset of traffic. For example, if only certain
DNIS/realms or service provider customers need to be recorded, by sending only that traffic to
the Recorder you can minimize processing overhead for the Recorder, thereby lowering hardware
costs.
• using this method minimizes the amount of network engineering required to access traffic, since it
can be routed over IP rather than through the use of SPAN/mirror ports or network tap devices.
This can reduce initial and ongoing network configuration costs and complexity.
Below is an example of Oracle SBC deployment in which it terminates a SIP trunk coming in from an
Internet Telephony Service Provider (ITSP), and relays the SIP/RTP traffic to an IP PBX in the contact
center. When the SRR feature is configured on the SBC, it replicates the SIP and RTP traffic to the
Recorder server as shown in the diagram.
 

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Genesys SIP Server Integration Requirements

Requirements
The Verint Systems Field Services personnel should confirm that the customer has the following
Switch/CTI configuration. Please refer to Verint Systems’ Product Central for the latest information on
supported Switch/CTI combinations.
.

Specification Requirement

Switch Name & Model Genesys SIP Server

Supported Switch Versions 7.5 to 8.1 *

CTI Middleware Genesys Voice Platform SDK

CTI Middleware Versions 7.5 to 8.1 *

* The requirement for version 8.1 applies to VoIP Delivery solutions that use the Genesys Call
Recording method (that is, SIP Server/Media Server call forking rather than an SBC).

For SIP Trunk Recording you will require the Oracle SBC 3810/3820 or 4250/4500 series, or AudioCodes
Mediant SBC.

 Note that the use of options outside those recommended by Verint may have a knock-on effect
or otherwise impact the Verint solution.

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Genesys SIP Server Integration Limitations/Known Issues

Limitations/Known Issues
Unable to replay session of second agent in three-way
conference with Oracle SBC
In an environment with Genesys and Oracle SBC, conferences are internally bridged and as such a
configured correlation key will not match up to the second agent's CTI, making the session inaccessible
for replay. This is expected behavior.

Recording of Internal and Consultation Calls in VoIP Delivery

Internal or consultation call between monitored agents

• Calling Agent is not recorded in internal and consultation calls as the calling agent's audio is not
delivered by the Genesys Media Server, and only the audio of the Called Agent is delivered to the
Recorder.
• As a result, the calling agent's segment is recorded without audio.
• To hear the audio of the call, one needs to listen to the recorded segment of the Called Agent.

Recording of Internal and Consultation Calls in VoIP Delivery

Internal or consultation call between monitored agents
• Calling Agent is not recorded in internal and consultation calls as the calling agent's audio is not
delivered by the Genesys Media Server, and only the audio of the Called Agent is delivered to the
Recorder.
• As a result, the calling agent's segment is recorded without audio.
• To hear the audio of the call, one needs to listen to the recorded segment of the Called Agent.

Internal or consultation call between monitored agent and non-monitored agent

• If the Called Agent is not monitored, Genesys Media server still delivers the audio to the
Recorder.
• As there is no CTI segment to match it to, the audio is recorded based on SIP signaling and it is not
associated with the CTI segment recorded for the calling agent.
• The audio of the Calling Agent is not recorded because the audio is not delivered to the
Recorder.

SIP Trunk Recording with Oracle SBC

Oracle SIP Trunk recording has the following limitations:

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Genesys SIP Server Integration Limitations/Known Issues

• IP Analyzer does not support RFC 2003 interception, therefore this release does not support
integration with Oracle SBC through IP Analyzer.
• For correlation to CTI in environments with multiple SIP servers, customization is required on the
SBC and in Genesys in order to maintain a proper correlation value throughout the life of the call.
• Each Recorder configured for Oracle SBC SIP Trunk Recording
• must be dedicated exclusively to the Oracle SBC SIP calls. These Recorders cannot be configured
to do additional IP Interception or IP Delivery Recording (however, you may perform TDM or
Screen Recording on the same server).
• must be dedicated to calls from a single Genesys server. If calls associated with several different
Genesys servers are required, then at least one Recorder per Genesys server is required.

SIPREC
The following apply only to SIPREC selective recording environments:
• Performance and Liability modes are not supported in SIPREC selective recording. If CTI is not
received for the call, no recording will happen.
• N+N is not supported in SIPREC delivery (only 1+1 for Recorder Integration Service redundancy,
and N+M all shared for Recorder redundancy).

Genesys Recording Indication Feature

In a scenario with 2 or more monitored extensions from the same PBX, all monitored extensions send
their INUM to the same CallID in Genesys and the order cannot be guaranteed as there is a race condition
between the update requests.

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Genesys SIP Server Integration Configuration Checklist

Configuration Checklist
Done? Task      

  1 Ensure all licensing requirements are met.

Related information 
Licensing, page 26

  2 Obtain all required information from the customer and validate site configuration.

Related information 
Site Preparation Checklist

  3 Configure the switch/third-party devices and Recorder according to your recording

type:

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Done? Task      

  VoIP Station-side VoIP Station-side SIP Trunk Delivery - SIP Trunk Delivery -
Interception Delivery SRR SIPREC
a. Configure the a. Configure the a. Configure a. Configure
switch for VoIP switch for VoIP Configure Oracle Configure
Station-side Station-side SBC for SRR, AudioCodes or
Interception, Delivery, page 27. page 29. Oracle SBC for
page 27. b. Complete general b. Complete general SIPREC, page 29.
b. Complete general Recorder Recorder b. Complete general
Recorder Configuration, Configuration, Recorder
Configuration, page 30. page 30. Configuration,
page 30. c. Configure the b. Configure the page 30.
c. Configure the Recorder for VoIP Recorder for VoIP c. Configure the
Recorder for VoIP Station-side Gateway Recording, Recorder for
Station-side Delivery, page 32. page 32. Configure SIPREC,
Interception, The use of full-time c. Configure the page 35.
page 30. vs. selective Recorder. The use of full-time
d. Create an recording will be d. Create an vs. selective
adapter, under CTI determined by the adapter, under CTI recording will be
Adapters, page 38. Recorder Type, Full Adapters, page 38. determined by the
Delivery (External Recorder Type, Full
Controlled) or Delivery (External
Selective Delivery Controlled) or
(Duplicate Selective Delivery
Streamed) (Duplicate
respectively, that Streamed)
you select in the respectively, that
member group. you select in the
d. Create an member group.
adapter, under CTI d. Create a Generic
Adapters, page 38. SIPREC adapter plus
at least one
additional adapter
to handle CTI. See
an adapter, under
CTI Adapters,
page 38.

Note that in Selective Recording and Delivery environments you must configure both the Genesys
SDK adapter and the SIP Proxy adapter, and associate them with the same data source.

Related information 
Recorder Configuration and Administration Guide

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Done? Task      

  Complete Site Acceptance Tests.

Related information 
Site Acceptance Tests

  Place a test call to an extension that is set to Record (either because of an extension
recording mode or Business Rule). If the call is not recorded or can’t be replayed,
troubleshoot.

  Configure Screen Recording (if required).

Related information 
Recorder Configuration and Administration Guide

  Configure Redundancy.

Related information 
High Availability for Voice and Screen, page 67

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Licensing
In Genesys environments, the customer requires the following licenses:
• Genesys Platform SDK License. This license is required for use of the Genesys Platform SDK or the
Voice Platform SDK, which is part of the Platform SDK solution.
• Agent licenses. When using the Genesys SDK Adapter, you need one license per Agent/DN
configured in the system.
• Genesys Recording Connector License which is necessary for all solutions except Delivery.
• Genesys Active Recording Connector License. This license is only required for Station-side Delivery
solutions.

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Configuration
To configure a deployment with Genesys IP Switch with the Genesys T-Server, first complete the tasks in
Switch Configuration, page 27, for all Genesys environments.
In Genesys SIP server recording (both VoIP Delivery and VoIP Interception), supervisors can monitor
agent calls. In this release, the Recorder Integration Service tracks connections on a call in such a way that
the supervisor is not interpreted as a tracked party/connection on the call. The specific adapters that
track call observers are the Genesys SDK and T-Lib. Parties marked as observers (the supervisor in this
case) are now ignored - if the party's role changes in the midst of a call, the tracking will account for this.
Next, refer to the following sections for Recorder configuration instructions, according to the specific type
of recording you want to perform:
• VoIP Station-side Interception, page 30
• VoIP Station-side Delivery, page 32
• VoIP Gateway Recording, page 32
Finally, for each of these deployments you must also configure the appropriate adapter (see CTI
Adapters, page 38).

 A note about hunt groups. If you configure hunt groups as part of the phone data source, do not
configure agent group/hunt group extensions as phone extensions in any other places, otherwise
you will not receive AgentLoggedOn/Off events from the CTI switch.

Switch Configuration
The Genesys adapter connects to the Genesys server over the LAN using TCP/IP via the T-library SDK. The
Genesys server in turn connects to the Genesys switch. The Genesys server provides the adapter with
real-time telephony information from monitored position IDs, DNs, Controlled DNs, and ACD Queues.
Information contained in these messages is tagged along with the recording.

VoIP Station-side Interception

VoIP Interception is VoIP recording using a SPAN port/port mirror. With VoIP Interception, spanning can be
accomplished from one of three general locations within a customers network topology. These locations
are at the gateway, switch-side (infrastructure), and station-side. With gateway recording, the VoIP traffic is
spanned at a gateway typically located at the point where the call enters the customer’s environment.
Switch-side (infrastructure) recording is accomplished by spanning the traffic on the station side of the
switch. Station-side recording is accomplished by spanning the data switch that the agent phones are on.

VoIP Station-side Delivery

The Genesys Media Server provides Media Server Markup Language (MSML)-based media services. Call
recording through MSML is provided through an external recording server such as Genesys Quality
Management, or a third-party voice recorder, i.e. the Verint Recorder. (It’s one of two available methods
that can be used with the GVP for VoIP delivery, the other being NETANN-based call recording, but for

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this integration you must use MSML-based call recording.) For a detailed explanation of how to configure
this feature, see Genesys’ Active Recording Ecosystem Solution Guide.
In addition, you may wish to consult Genesys’ documentation (at
http://docs.genesyslab.com/Documentation/SIPS) for the following:
• Call Recording in a Genesys SIP environment: The configuration requirements to enable Call
Recording in a Genesys SIP environment are explained in detail in the Call Recording-MSML-
Based section of the Genesys Framework SIP Server Deployment Guide.
• Integrate the Genesys SIP Server with the Genesys Media Server: For an explanation on
integrating the Genesys SIP Server with the Genesys Media Server, refer to the Configuring
Genesys Media Server section in the Genesys Framework SIP Server Deployment Guide.

Configure a Verint Recorder as a third-party recording server

1 Create a new Resource Access Point object.
2 Open the Options tab of the object and ensure that a gvp.rm section exists.
3 Open the gvp.rm section and set the aor to point to the Integration Service's IP and port, for
example: sip:10.165.141.124:5060.
 

 The Integration Service's SIP Proxy adapter serves as a SIP endpoint and not the IP Recorder.

4 Open the provision section and set recording-server to 1.

Configuring the Media Control Platform of GVP

1 Open the Options tab of the MCP object and ensure that the vrmrecorder section exists.
2 Open the vrmrecorder section and set the sip.routeset parameter to point to Resource Manager
(RM) in the following format:
<sip:[rm-ip]:[rm-port];lr>
 

 These settings only relate to VoIP Station-side Delivery recording. The signs < and > are both
required for MCP to send the recording request to RM.

Configure the Interactive Voice Response profile

1 Navigate to Voice platform > IVR profiles > Default application.
2 In the Options tab in the gvp.service-parameters, configure the following options:
• recordingclient.recmediactl = fixed,1
• Recordingclient.recdest = fixed,sip:[rm-ip]:[rm-port]
To record both parties on a call, the Genesys switch must be configured accordingly.
The default setting of the recmediactl parameter value is 2. Using this default setting, separate SIP
sessions are established on the recording server, one for each recording stream, and 2 INUMs are
recorded.
By setting the recmediactl parameter value to 1, a single SIP session is used for both streams (with
multiple m-lines in the Session Description Protocol [SDP]). This is the recommended configuration for
integration with RIS to ensure that one INUM is recorded including both parties.

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Configure Oracle SBC for SRR

This section applies only to Oracle SBC Session Replication. The Recorder emits a “No Packets Received”
alarm in order to alert Recorder administrators that there is an issue in the network. In order for this
alarm to be effective, the Recorder must always receive traffic on the RFC 2003 Interface. The
recommended way to achieve this is to enable a SIP OPTIONS ping-method between the SBC and its far-
end SIP trunk for the realms that are assigned to call-recording-servers. You can configure this in the
session-agent record in the SBC configuration. The ping-method must be set to OPTIONS, and ping-
interval must be less than 30 (15 is recommended).

Example:  
acmesystem(session-agent)# show

session-agent

hostname pstn-out-rec1-sa

ip-address 172.29.101.11

port 5060

state enabled

app-protocol SIP

app-type

transport-method UDP

realm-id pstn_out_rec1_realm

ping-method OPTIONS

ping-interval 15

Configure AudioCodes or Oracle SBC for SIPREC

To use either AudioCodes or the Oracle SBC for SIPREC,
• the SBC must be set up to pass a Genesys call GUID in the SIPREC metadata.
• Genesys must be set up to pass the Genesys call GUID in CTI.
• a static Genesys GUID is required in both SIPREC metadata and Genesys CTI in order to support
cradle-to-grave tracking of calls. In environments with multiple Genesys SIP Servers, this may
require additional customization on the SBC and in the Genesys SIP environment.

Secure RTP with SIP over TLS

SIPREC deployments support TLS encryption for SIP (SIP/TLS) signalling, and Secure Real-time Transport
Protocol (SRTP) for audio. The SIPREC adapter will accept public/private keys stored in either JKS or PKCS12
format. The key algorithm must be RSA.

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To configure your system for SIP over TLS/SRTP you must do the following as part of your overall VoIP
Gateway recording configuration:
• Set up certificates as described in Set Up Certificates, page 82.
• Create a Create a SIPREC Adapter, choosing SIP over TLS as the SIP Protocol, and completing the
TLS Parameters section.

Recorder Configuration
Complete the following steps for all recording types.
1 Install and configure the Recorder for the Enterprise, using the following documents:
• Installation Guide
• Enterprise Manager Configuration and Administration Guide
• Recorder Configuration and Administration Guide
2 Associate the IP Recorder with the Recorder Integration Service Role.
3 In Enterprise Manager, click System Management > Settings.
4 Select a Recorder from the left-hand pane.
5 Click the Server Roles tab.
6 Select the check box beside the name of the appropriate recording-related role.
7 Click Save. Wait for items in the Configuration Status queue to clear.
8 Restart your system:
a. Select the Recorder, then the Settings tab.
b. Click Launch to start Recorder Manager.
c. Click Operations > Start and Stop.
d. Click Reboot.
See the Enterprise Configuration Guide for further details.
e. Create a Phone data source in Enterprise Manager, and select Genesys as the Switch/Sub Type.
 

Next, complete one of the following sections to configure the Recorder, depending on your environment:
• VoIP Station-side Interception, page 30
• VoIP Station-side Delivery, page 32
• VoIP Gateway Recording, page 32

VoIP Station-side Interception

Use the following procedure to configure the Recorder for VoIP Station-side Interception.
1 In Recorder Manager, click General Setup > Capture Settings > Cards and Filters.
2 Set the Recording Type of the network interface card used for recording to Interception.
3 Click the Protocols tab and make sure that SIP is enabled.
4 In Enterprise Manager, under the Genesys data source create an IP Extension Pool member group
for the extensions you want to record.
5 Set the Recorder Control Type to Recorder Controlled.

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Configure the SIP Proxy IP Address values

SIP protocol messages do not include the call direction/type (incoming call or outgoing call) information.
The IP Capture engine requires this information to properly tag direction and extension for SIP Calls. The
Recorder uses SIP Proxy IP Addresses to identify the direction; therefore, you must configure the SIP
ProxyIPAddresses parameter for VoIP Station-side Interception.
1 Navigate to the Impact360\Software\ContactStore folder within the installation directory.
2 Open the IPCaptureConfig.xml file in a text editor such as Notepad.
3 Locate the following section:
<x:SIP>

4 Within the <x:SIP> tags, configure the ProxyIPAddress section with addresses.
The following example illustrates the modified configuration for two proxy addresses:
<x:ProxyIPAddresses>
<x:ProxyIPAddresses>
<x:ProxyIPAddress>100.10.10.10</x:ProxyIPAddress>
<x:ProxyIPAddress>100.10.10.11</x:ProxyIPAddress>
</x:ProxyIPAddresses>

5 Save and close the file.

6 Restart the Recorder IP Capture service.

Configure the UseInviteURI

In Genesys environments where incoming calls contain an extension in the INVITE URI rather than in the
To header, and outgoing calls contain an extension in the Contact header rather than in the From
header, you need to set "UseInviteURI" parameter in the IPCaptureConfig.xml file to true in order to
extract the extension from InviteURI/Contact header.
1 Navigate to the Impact360\Software\ContactStore folder.
2 Open the IPCaptureConfig.xml file.
3 Configure the UseInviteURI to true.
Example:
<x:UseInviteURI>true</x:UseInviteURI>
4 Click Save.
5 Restart the RecorderIPCaptureEngine Service.

 For IP Analyzer, update the IPAnalyzerConfig.xml file and restart the Recorder Analyzer Service.

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VoIP Station-side Delivery

Use the following procedure to configure the Recorder for VoIP Station-side Delivery.

Before you begin 

A certain number of User Datagram Protocol (UDP) ports must also be made available for Delivery. The
total number depends on the number of licenses you have, and on the requirements of a given
integration. Calculate your port requirements per Recorder using the formula outlined in "Calculating the
number of ports required" in the VoIP Delivery Deployment Reference Guide.
1 In Enterprise Manager, under the Genesys data source create an IP Extension Pool member group
for the extensions you want to record.
2 Set the Recorder Control Type:
• For Selective Recording, set to Duplicate Streamed.
• For Full Recording, set to External Controlled.
3 In Recorder Manager, click General Setup > Capture Settings > Cards and Filters.
4 Set the Recording Type of the network interface card used for recording to Delivery and enable No
Configured Protocols.
5 Verify that the Starting Port and Ending Port of the network interface card used for recording have
been set. This defines the range of ports that are allocated for audio delivery. In a full recording
environment, where ports are allocated for recording upon Integration Service startup, each
extension must have a pair of ports allocated for it. Refer to the formula described at the top of this
section to calculate the total number required.
6 Add all of the extensions to be recorded to this member group.

 For local recording at the extension location, select the member group used to record the desired
extensions specific to that location.

VoIP Gateway Recording

There are three ways to implement VoIP Gateway Recording.
The first two are types of SIP Trunk Delivery: using Oracle SBC for SRR, or using the Genesys SIP Server
itself to provide SIP invite messages (with correlation information) to the Recorder Integration Service.
If your network topology is such that the Genesys SIP server cannot provide extensions as part of SIP
signaling, you can correlate CTI to SIP signalling using a correlation key (rather than an extension).
Another case in which recording is based on a correlation key rather than an extension would be that of
an extension that is reported as part of a SIP INVITE, but reported with an incorrect value due to site
topology. This can occur, for example, when a call is routed through multiple T-Servers.
In other instances, a correlation key can be used to support the recording of IVR calls that don’t have CTI.
These calls will be tagged in the database with the correlation key, and once the call reaches an agent it will
be stitched to the IVR segment using that same correlation key.
A third type of VoIP Gateway is SIPREC, wherein a session replicating device like the AudioCodes Gateway
or SBC acts as an SRC between the Recorder Integration Service and the SIP Trunks/SIP phones/call center
system.
Use one of the following procedures, depending on your method of implementation:
• Configure SRR via Oracle SBC, page 33

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• Configure SIP Trunk Delivery via Genesys SIP Server, page 33

• Configure SIPREC, page 35

Configure SRR via Oracle SBC

For an Oracle SBC Integration you must use the following settings when configuring the IP Recorder.
Note that SIP Trunk Recording is established at the member group level (not the extension level).
1 In Recorder Manager, select the IP Recorder then click General Setup > Capture Settings > Cards
and Filters. Set the Recording Type to RFC 2003 Interception.
2 In Enterprise Manager:
a. Create a Genesys Phone data source and configure one of the following Server Types under
Settings > Device IP Configuration:
- For switch signalling interfaces, configure PBX Side - Near End as the server type, and define
either its IP address or host name.
- For SIP trunk recording, configure PSTN Side - Far End as the server type, and define either
the IP address or host name of the call center’s SIP Trunk Interface server.
b. Within the Phone data source, create a Gateway Side Correlation Pool member group.
 

 Do not include a Correlation Key in the member group settings unless instructed to do so by
Verint. The system will use the correct setting by default, so leave this field blank. See “Gateway
Side Correlation Pool Settings” the Recorder Configuration and Administration Guide for additional
details.
3 In a text editor such as Notepad, open the files IPCaptureConfig.xml and, optionally,
IPAnalyzerConfig.xml (located in %IMPACT360SOFTWAREDIR%\ ContactStore). Locate
<x:CustomSIPTags> under the tag <x:SIP>. Within <x:CustomSIPTags>, add <x:CustomSIPTag> with
the Genesys ID, as indicated in bold in the following example.

<x:CustomSIPTags>
<x:CustomSIPTag Overwrite="true" Section="SIP
Header">X-Genesys-CallUUID</x:CustomSIPTag>
</x:CustomSIPTags>
In addition:
• SBC must use port 5060; this is the default port so there is no need to configure this (but
conversely, it must not be changed).
• The Network Interface Card (NIC) requirements for this integration are the same as those cited in
the Customer Furnished Equipment Guide under “Requirements for RFC 2003." For RFC 2003
Interception you can have multiple NIC cards, but it is possible to configure the Recorder to use
only one NIC.
To learn more about the redundancy that Oracle SBC provides, see Oracle SBC Redundancy, page 74.

Configure SIP Trunk Delivery via Genesys SIP Server

In this method, Genesys SIP Server/Media Server call forking is used for Delivery, rather than an SBC.

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1 To ensure accurate speaker identification (optional) when using speaker separation, in Enterprise
Manager enter the IP address or host name of the call center’s SIP trunk interface in the data source
for the PSTN Side - Far End server type under Settings > Device IP Configuration.
2 In Recorder Manager, click General Setup > Capture Settings > Cards and Filters. Set the
Recording Type to High Capacity Delivery.
3 Under General Setup > Capture Settings > Protocols, verify that no protocols are selected.
4 Configure Custom SIP Correlation Keys. This step is optional. If you need to use a correlation key
other than the defaults (which are SIP:partyid and CTI: GSIP_REC_FN from the UserData section of
the SIP invite messages), configure the custom keys as follows.

 In both of the following cases, use names that are different from the values configured in
IPCaptureConfig.xml. Do not use the same names as those of the Recorder attributes.

• CTI correlation key—Create an attribute in Enterprise Manager to hold each CTI attribute that
will be used for correlation, as described in the “Create or Edit an Attribute” section of the Recorder
Configuration and Administration Guide.
• In the Genesys SDK adapter, map the CTI correlation attributes to their External Names as received
in CTI, as described in the “Map Attributes to an Adapter” section of the Recorder Configuration
and Administration Guide.
• Recorder correlation key—Create an attribute in Enterprise Manager to hold each Recorder
attribute that will be used for correlation, as described in the “Create or Edit an Attribute” section
of the Recorder Configuration and Administration Guide.
• In the SIP Proxy adapter, map the Recorder correlation attributes to their External Names as
received in the SIP invite, as described in the “Map Attributes to an Adapter” section of the
Recorder Configuration and Administration Guide.

 If you manually configure keys, they will override the defaults. For this reason, if you intend to
use the default(s) as well, you must manually configure these in addition to the other keys you
are adding.

5 Include the correlation key attributes established above in your “Gateway Side Correlation Pool
Member Group Settings” (under Correlation Key), as described in the Recorder Configuration and
Administration Guide.
Example:  
Consider the following CTI and SIP invite messages.
From CTI,
Dispatching Event CTIEvent<si.19> --> <CallTracker> Size<0>
Int<AdapterId> = 19 ; Str<AdapterName> = Genesys SDK - SIP Server ; Int<SwitchId> =
252 ; Str<SwitchName> = Genesys - Test
Folder<event>
Int<Event> = 85 ; Str<EventStr> = EventAttachedDataChanged ; Int<PropagatedCallType> =
1 ; Int<CallType> = 1 ; Str<CallUuid> = 3QT8ORSUBL46R4S6IEVQOOFHRC000034
Int<ProtocolId> = 5415651 ; Str<DNIS> = 70052 ; Long<EventSequenceNumber> = 24357 ;
Long<ConnID> = 44194116344475743 ; Str<ConnIDStr> = 009d02508ddff05f
Int<CallID> = 100
Folder<ThisParty>
Int<DNRole> = 1 ; Str<DN> = 70053
Folder<UserData>

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Str<GSIP_REC_FN> = 3QT8ORSUBL46R4S6IEVQOOFHRC000034 ; Str<GSIP_RECORD> = ON

Folder<ThirdParty>
Str<DN> = 70052

From the SIP INVITE,

Message<Event> callStateMessage<SIPProxy> --> <CallTracker>
Int<AdapterId> = 16 ; Str<AdapterName> = Genesys - SIP Proxy ; Int<SwitchId> = 252 ;
Str<SwitchName> = Genesys - Test
Folder<event>
Folder<header>
Str<partyid> = 3QT8ORSUBL46R4S6IEVQOOFHRC000034
Folder<to>
Str<OtherDN> = 70053 ; Str<recordDN> = 70052 ; Str<CallUUID> =
3QT8ORSUBL46R4S6IEVQOOFHRC000034 ; Str<DN> = 7005
 
To use DNIS as the CTI correlation key, and recordDN as the SIP (Recorder) correlation key, you would
configure the attributes as follows. (Note that the attributes are case-sensitive.)
• Map the Genesys SDK adapter attribute, created to be used as the CTI correlation key, to
event.DNIS.
• Map the SIP Proxy adapter Recorder attribute, created to be used as the Recorder correlation key,
to event.header.to.recordDN.
 

Configure SIPREC
Use the following procedure to configure the Recorder for SIPREC via Oracle SBC or an AudioCodes
Gateway.

 Internal calls are not recorded when deploying a gateway recording solution in a Genesys
integration.

1 In the Recorder Manager, select the IP Recorder then click General Setup > Capture Settings >
Cards and Filters, then set the Recording Type of the network interface card used for recording to
Delivery, for both full-time and selective SIPREC.
2 In Enterprise Manager, create a Gateway Correlation Pool member group and set the Recorder
Control Type as follows:
• For full-time recording, select External Controlled.
• For selective recording, select Duplicate Streamed.
Note that for SIPREC a data source may have a mix of member groups for both full-time (set to
External Controlled) and selective (set to Duplicate Streamed). In these cases, recording will be
determined based on the configuration of the member groups. You should therefore configure the
member groups so that the there's a clear distinction in terms of what should be recorded where (you
can do this, for example, by using location IPs).
3 Define the phones to monitor on the data source and associate the member group with the relevant
Recorders under Shared Recorders.

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 Do not include a Correlation Key in the member group settings unless instructed to do so by Verint.
The system will use the correct setting by default, so leave this field blank. See "Gateway Side
Correlation Pool Settings" in the Recorder Configuration and Administration Guide for additional
details.

4 In Recorder Manager, set the Recording Type of the network interface card used for recording to
Delivery.

Certificates for Genesys Platform SDK and Interaction Server

When the Genesys T-Server is configured in secure mode, you should configure Verint's Genesys SDK
adapter to use a TLS connection. This will allow encrypted CTI event messages to be transported from the
Genesys Platform SDK or Genesys Interaction Server to the Verint platform, where they will be
decrypted.
There are a couple of different ways to configure certificates and authentication, depending on your
setup.

Certificates

Configuration using a custom JKS file

Create a new JKS key store file to hold the client certificate, which will be used for the key manager for
validation.
Specify the path for the keystore and its password when you configure the adapter. The password for the
JKS file and the client private key password must be the same.
See: "Create new keys using a self-signed certificate" or "Create new keys using a CA-signed certificate" in
Set up keys for certificates, page 90.

Configuration using Verint default certificates

You can use Verint’s existing server certificate as client certificate for secured communication with
Genesys.
See: "Use existing keys" in Set up keys for certificates, page 90.

Authentication
Whether you need to provide a certificate to Genesys for authentication depends on whether you are
using 'mutual' authentication on the Genesys side.

Mutual authentication
If the Genesys server requires mutual authentication you must provide it with a certificate that is trusted
by Genesys.
This can be either included in the custom JKS or, if SSL is configured on the server, using Verint standard
client certificates from IMPACT360/Conf/Security/cert.pem.
See Provide a certificate to Genesys, page 88.

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Non-mutual authentication
If you are not using mutual authentication, skip the tasks under Provide a certificate to Genesys, page 88.

Workflow
To set up TLS for Genesys, do the following:
• Set up keys for certificates, page 90
• If you are using mutual authentication on the Genesys side, Provide a certificate to Genesys,
page 88.
• Configure the adapter for TLS, page 89

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CTI Adapters
The Genesys adapter connects to the Genesys server over the LAN using TCP/IP via the Genesys SDK. The
Genesys server provides the adapter with real-time telephony information from monitored extensions,
hunt groups, and VDNs. Information contained in these messages is tagged along with the recording.
• The Genesys Voice Platform SDK Adapter. This adapter is recommended for use in Version
11. You must define this type of adapter if using Genesys SDK version 8.0.
• Genesys (Legacy T-lib) Adapter. This adapter supports Genesys systems where the T-lib
adapter is already in use or where the SDK is not licensed.

 For Selective Recording and SIP Trunk Recording, you must configure both the Genesys SDK
adapter and the SIP Proxy adapter, and associate them with the same data source.
To record SIPREC calls, you must create at least two adapters: a Generic SIPREC adapter
(described below) and at least one additional adapter to handle CTI.
Create at least one of the following adapters in Recorder Manager:
• Create a Genesys Voice Platform SDK Adapter, page 39
• Create a Genesys (Legacy T-lib) Adapter, page 41
• Create a Genesys SIP Proxy Adapter, page 1
• Create a SIPREC Adapter
To create an adapter, launch the Recorder Manager and click General Setup > Integration Adapters >
Settings > Create. Each requires the following settings:

Field Description

Adapter Name Type a unique name for this adapter. Do not use any special
characters or characters that truncate xml. This field is not case
sensitive.

Description Type a description of the adapter (optional).

Adapter Type A read-only field that lists the adapter selected above.

Startup Type Select a startup type: Automatic, Manual, or Disabled.

DataSource You can have more than one adapter in your enterprise, and you
can associate multiple adapters with a single data source.
However, a single data source should be assigned only one main
CTI adapter (not tagging only). A primary CTI adapter is defined
as having primpary control control. SIP proxy adapters are not
considered CTI adapters.

Next, proceed to the appropriate section below to learn more about the settings specific to that adapter.

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Genesys SIP Server Integration Configuration

Create a Genesys Voice Platform SDK Adapter

1 Choose General Setup > Integration Adapters > Settings.
2 Click Create, then select Genesys SDK Adapter.
3 Specify settings for this adapter in the right-hand pane, referring to the following table (all fields are
required unless otherwise indicated):

Field Description

T-Server Hostname Type the name of the primary Genesys server (set to Verint by
default).

T-Server Port Specify the port on which the primary T-Server listens for connections
from the clients. 9001 is the default.

Application Name This value is used to identify the Recorder Integration Service in the
Genesys logs. It should be a unique value that describes the recording
system/server. Verint Recording is the default.

Use Secondary Select this check box to use a failover server.

Server for failover

Secondary T-Server Type the name of the secondary Genesys server (set to verint by
Hostname default). This field is required if you select the Use Secondary Server
for failover check box.

Secondary T-Server Specify the port on which the secondary T-Server listens for
Port connections from the clients. 9001 is the default. This field is required
if you select the Use Secondary Server for failover check box.

Get RTP Information Genesys will provide RTP information if it used with a Cisco switch or
Genesys SIP server. To obtain this information you must select this
check box. The option is not enabled by default. Only required for CTI
Controlled recording.

RTP Password If you have enabled Get RTP Information, type the password
configured in Genesys.

Unique Global Call Used to associate multiple calls together in a consult, conference,
ID transfer or other multi-call scenario. Select one of the following:
NetworkCallID/NetworkNodeID
CallID
ConnID
Other
Note: Although NetworkCallID/NetworkNodeID is an option for
this field, there is a known issue with Genesys SDK version 8.0 where
this ID won’t work as the Unique Global Call ID.

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Genesys SIP Server Integration Configuration

Field Description

Tagging Only If this adapter is used in addition to any other call tracking adapters
on the same data source and this adapter is expected to only Tag CTI
data for the calls that are tracked by another adapter.
See Default Tagging by the Tagging Only Adapter, page 41.

Process Agent Event This option will enable the Genesys adapter to process agent state
changes, specifically Agent Login and Logout.

Event Based After This option will enable the Genesys adapter to stop screen recording
Call Work based on after call work events, specifically Agent Ready and Logout.
Off by default.

4 Under Advanced Settings, use the Key and Value fields to enter any proprietary pairs that are in
use in your system.
5 Click Save. The adapter appears in the left-hand pane.
6 Select the adapter in the left-hand pane and click the Start button to start the adapter.
7 Under the Send Recording Indication to Genesys pane, enable the Recording Indication Feature
(off by default) by configuring the following parameters.
 

Field Description

Send Recording Select this checkbox to enable the Genesys Recording Indication
Indication to Feature.
Genesys The Genesys Recording Indication feature prompts the Recorder
Integration Service to send a recording identifier back to Genesys for
all recorded calls. The Genesys Servers exchange call context with all
the KVPs for calls that span more than one Server. If the Genesys
Recording Indication feature is enabled, it allows customers to keep
recording calls that are transferred across separate PBXs.
This feature uses the AttachUserData request of the Genesys
interface to provide the Genesys ConnectionID for the call that is
updated, and also provides one or more of the following:
• KVP name of the recording status key field and its configured
recording status value to update.
• KVP name of the primary recording INUM and the INUM to
update.
• KVP name of the Contact ID and the Contact ID to update.
This feature supports any recording environment with Genesys call
tracking adapter or Genesys tag-only adapters.

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Genesys SIP Server Integration Configuration

Field Description

Recording Status Enter the Genesys KVP field name (for example Recorded) that
Key Name contains the key of the recording status (compatibility with v10).
The Recording Status Key Name field is empty by default.

Recording Status Enter the Genesys KVP field name (for example True) that will be
Value tagged as a recording status (compatibility with v10).
The Recording Status Value Name field is empty by default.

Primary Recording Enter the Genesys KVP field name that will be updated with the
INUM Key Name primary recording INUM.
The default is VerintPrimaryRecordingInum.

Contact ID Key Enter the Genesys KVP field name that will be updated with the contact
Name id of the primary recording INUM.
The default is VerintContactId.

Default Tagging by the Tagging Only Adapter

Tagging Only adapters must be used in conjunction with another CTI adapter. All standard attributes
come from the controlling CTI adapter, while those from the Tagging Only adapter are treated as custom
data.
The Genesys SDK or Legacy T-Lib Tagging Only adapter only processes Agent states (Agent Login, Agent
Logout, and so on) and custom attributes, while the other CTI adapter provides information to track calls,
recording, and some level of tagging. If you want tagging for any additional attributes, use custom data to
capture them as described in the Recorder Configuration and Administration Guide.

Create a Genesys (Legacy T-lib) Adapter

The Genesys (Legacy T-lib) Adapter is EOL (End of Line) and the Genesys SDK adapter should be used
 
instead.

Define a Genesys (Legacy T-lib) Adapter if using Genesys T-Server version 7.2 or earlier.
1 Choose General Setup > Integration Adapters > Settings.
2 Click Create, then select Genesys (Legacy T-lib) Adapter.
3 Specify settings for this adapter in the right-hand pane, referring to the following table (all fields are
required unless otherwise indicated):

Field Description

Adapter Name Type a unique name for this adapter. Do not use any special
characters or characters that truncate xml. This field is not case
sensitive.

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Field Description

Description Type a description of the adapter (optional).

Adapter Type A read-only field that lists the adapter selected above.

Startup Type Select a startup type: Automatic, Manual, or Disabled.

DataSource Select a data source.

T-Server Hostname Type the name of the primary Genesys server (set to Verint by
default).

T-Server Port Specify the port on which the primary T-Server listens for connections
from the clients. 9001 is the default.

Application Name This value is used to identify the Recorder Integration Service in the
Genesys logs. It should be a unique value that describes the recording
system/server. Verint Recording is the default.

Use Secondary Select this check box to use a failover server.

Server for failover

Secondary T-Server Type the name of the secondary Genesys server (set to verint by
Hostname default). This field is required if you select the Use Secondary Server
for failover check box.

Secondary T-Server Specify the port on which the secondary T-Server listens for
Port connections from the clients. 9001 is the default. This field is required
if you select the Use Secondary Server for failover check box.

Get RTP Information Genesys will provide RTP information if it used with a Cisco switch or
Genesys SIP server. To obtain this information you must select this
check box. The option is not enabled by default. Only required for CTI
Controlled recording.

RTP Password If you have enabled Get RTP Information, type the password
configured in Genesys.

Unique Global Call Used to associate multiple calls together in a consult, conference,
ID transfer or other multi-call scenario. Select one of the following:
NetworkCallID/NetworkNodeID
CallID
ConnID
Other

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Genesys SIP Server Integration Configuration

Field Description

Tagging Only If this adapter is used in addition to any other call tracking adapters
on the same data source and this adapter is expected to only Tag CTI
data for the calls that are tracked by another adapter.

Process Agent Event This option will enable the Genesys adapter to process agent state
changes, specifically Agent Login and Logout.

Event Based After This option will enable the Genesys adapter to stop screen recording
Call Work based on after call work events, specifically Agent Ready and Logout.
Off by default.

4 Under Advanced Settings, use the Key and Value fields to enter any proprietary pairs that are in
use in your system.
5 Click Save. The adapter appears in the left-hand pane.
6 Select the adapter in the left-hand pane and click the Start button to start the adapter.

SIP Proxy
Use the following procedure to create a SIP Proxy adapter.

Procedure 
1 Click General Setup > Integration Adapters > Settings.
2 Click Create, then select SIP Proxy Adapter.
3 Specify settings for this adapter in the right-hand pane, referring to the following table (all fields are
required unless otherwise indicated):

Field Description

Adapter Type a unique name for this adapter. Do not use any special
Name characters or characters that truncate XML. This field is not case
sensitive.

Description Type a description of the adapter (optional).

Adapter A read-only field that lists the adapter selected above.

Type

Startup Select a startup type: Automatic, Manual, or Disabled.

Type

DataSource Select the LiveVox data source you created previously.

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Genesys SIP Server Integration Configuration

Field Description

Record Select this check box to record any device, even if it’s unrecognized.
Unknown Note that for LiveVox you should not disable this check box, as it may
Devices result in missed recordings for secondary agents who join a
conference with an unrecognized ID.

SIP Protocol Select the protocol that the SIP Proxy adapter will use.
• SIP over UDP—User Datagram Protocol (UDP) has a lower
overhead but will result in more recording loss in the case of
failures.
• SIP over TCP—Transmission Control Protocol (TCP) is
recommended as it is a more reliable protocol providing
earlier detection of failures.
• SIP over TLS—Transport Layer Security (TLS) is designed to
secure client-server communications over a network. It
operates on segments of network connections above the
Transport Layer.

Listen at IP Specify the local machine’s NIC IP Address. If Enterprise Manager is

Address installed on the Integration Service machine, you must use a port
other than 5060 as the listen port.

Port Type the port for the IP address specified above.

Note: The specified port cannot be in use by any other application or
adapter on the local server.

Supported Codecs
Select a supported codec from the list of those available.

TLS Parameters
Complete the settings in this section if you are using Secure RTP with SIP over Transport Layer Security
(TLS).

Field Description

KeyStore To use the standard keys that are installed with the recorder, leave this field set to
Filename the default. If you have generated a new key, type the path to that key here.

Password If you used the standard, pre-installed keys, enter the password provided in the
associated Recorder Integration Guide.

File format Select the file format that matches that of the file specified in the KeyStore Filename
field.

Support sRTP Select this check box to support fallback to RTP if secure/encrypted RTP is not
to RTP available.
fallback

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Genesys SIP Server Integration Configuration

4 Under Advanced Settings, use the Key and Value fields to enter any proprietary pairs that are in
use in your system.
5 Click Save. The adapter appears in the left-hand pane.

SIPREC Adapter
Use this adapter to support the retrieval of custom tags in SIP headers or SIPREC metadata in SIP Trunk
Recording.

Procedure 
Complete these steps to create a Generic SIPREC adapter.
1 Click General Setup > Integration Adapters > Settings.
2 Click Create, then select Generic SIPREC Adapter.
3 Specify settings for this adapter in the right-hand pane, referring to the following table (all fields are
required unless otherwise indicated):

Field Description

Adapter Type a unique name for this adapter. Do not use any special characters or characters
Name that truncate xml. This field is not case sensitive.

Description Optional: Type a description of the adapter.

Adapter A read-only field that lists the adapter selected above.

Type

Startup Select a startup type: Automatic, Manual, or Disabled.

Type

DataSource Select a data source.

SIPRec Specify the far-end SIPREC device being used for this integration.
Device
Type

SIP Select the protocol that the SIPREC adapter will use:
Protocol • SIP over UDP—User Datagram Protocol (UDP) allows applications to send
messages to hosts on an IP network. UDP has a lower overhead because it
assumes that error checking is either not required or that this processing is done
elsewhere.
• SIP over TCP—Transmission Control Protocol (TCP) is best for applications (such as
email and file transfer) that require a reliable data stream service.
• SIP over TLS—Transport Layer Security (TLS) is designed to secure client-server
communications over a network. It operates on segments of network connections
above the Transport Layer.

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Field Description

Listen at IP Specify the local machine’s NIC IP Address. The listening IP address and port specified
Address below must match the recording realm profile configured on the SBC.
If Enterprise Manager is installed on the Integration Service machine, you must use a
port other than 5060 as the listen port.

Port Type the port for the IP address specified above.

 The specified port cannot be in use by any other application or adapter on the
local server.

Supported Codecs
Select a supported codec from the list of those available.

4 Under Advanced Settings, use the Key and Value fields to enter any proprietary pairs that are in
use in your system.
5 Click Save. The adapter appears in the left-hand pane.
6 Click Restart.

Tag Custom Attributes Using the SIPREC Adapter

1 Refer to the procedure “Identify Required Third-Party CTI Attributes” in the Recorder Configuration
and Administration Guide. In this case, when you reach step 2, look for the following in the Integration
Service log file:
Folder<SIP>
Folder<HEADERS>
...
Folder<SDP>
...
2 From the messages in these sections you can start to identify the data you want to tag.
Example
In the following log sample, you can see that the messages contain a serial number (SerialNo) in
the RS_Source folder under Extension Data, and a string labelled User-Agent.
[IEMessage |0011|D] 2013/12/13 13:27:08.829 Dispatching Event recording<SIPProxy> -->
<CallTracker> Size<0>
[IEMessage |0011|H] 2013/12/13 13:27:08.840 Int<AdapterId> = 2 ; Str<AdapterName> =
siprec adapter ; Int<SwitchId> = 110 ; Str<SwitchName> = Genesys DS ; Str<dataMode> =
complete
[IEMessage |0011|H] 2013/12/13 13:27:08.859 Str<rSessionId> =
[email protected]
[IEMessage |0011|H] 2013/12/13 13:27:08.868 Folder<extensiondata>
[IEMessage |0011|H] 2013/12/13 13:27:08.878 Folder<RS_Source>

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Genesys SIP Server Integration Configuration

[IEMessage |0011|H] 2013/12/13 13:27:08.887 Str<type> = NetrixPhone ; Str<serialNo> =

0706001464
 
[IEMessage |0011|H] 2013/12/13 13:27:08.896 Folder<SIP>
[IEMessage |0011|H] 2013/12/13 13:27:08.907 Folder<HEADERS>
[IEMessage |0011|H] 2013/12/13 13:27:08.916 Str<Content-Length> = 2401 ; Str<Call-ID> =
[email protected] ; Str<CSeq> = 15 INVITE ; Str<Max-
Forwards> = 70
[IEMessage |0011|H] 2013/12/13 13:27:08.925 Str<From> = "SipTester"
<sip:[email protected]>;tag=581 ; Str<To> =
<sip:[email protected]:5080> ; Str<Require> = recording-session
[IEMessage |0011|H] 2013/12/13 13:27:08.935 Str<Via> = SIP/2.0/TCP
127.0.0.1:5095;branch=z9hG4bK-313933-
2599f2c035d072dd2c5c883a39ebd141;received=10.56.20.72;rport=57083 ; Str<User-Agent>
= Mediant
[IEMessage |0011|H] 2013/12/13 13:27:08.944 Str<Contact> = "12345"
<sip:[email protected]:5095;transport=tcp>;src ; Str<Content-Type> =
multipart/mixed;boundary=MultipartBoundaryMarker
[IEMessage |0011|H] 2013/12/13 13:27:08.954 Str<Route> =
<sip:10.56.20.72:5080;lr;transport=tcp>
[IEMessage |0011|H] 2013/12/13 13:27:08.963 Folder<SDP>
[IEMessage |0011|H] 2013/12/13 13:27:08.973 Str<v> = 0 ; Str<o> = sipXecs 5 9 IN IP4
10.221.14.222 ; Str<s> = phone-call ; Str<c> = IN IP4 10.221.14.100 ; Str<t> = 0 0
[IEMessage |0011|H] 2013/12/13 13:27:08.982 Array<MEDIA> [1]
[IEMessage |0011|H] 2013/12/13 13:27:08.991 [0]: Str<m> = audio 16396 RTP/AVP 8 ;
Str<ptime> = 20 ; Str<rtpmap> = 8 pcma/8000/1 ; Str<label> = streamlabel1
With this information in hand, you can map the attributes “Agent Name” and “ANI” to the SIPREC
adapter, allowing it to pick up the real values found in the IEMessages. (If you need to create custom
attributes, follow the procedure “Create or Edit an Attribute” in the Recorder Configuration and
Administration Guide first.)
a. In Recorder Manager, click System > Integration Adapters > Settings.
b. Follow the procedure “Map Custom Attributes to an Adapter” in the Recorder Configuration and
Administration Guide, specifying, in this example,
• SIP.HEADERS.User-Agent as the External Name for the Agent Name attribute. This
will enable tagging of the data from the User-Agent field identified in the sample log above.
• extensiondata.RS_Source.serialNo as the External Name for the ANI attribute. This will
enable tagging of the serial number identified above.
c. Click Save.

Related topics 
Supported Integration Service CTI adapters

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Genesys SIP Server Integration Advanced Deployments

Advanced Deployments
The following scenarios require specific configuration:
• Genesys IVR Recording, page 48
• Maintenance, page 51

Genesys IVR Recording

The GVP IP Communication Server (GVP IPCS) provides IVR capabilities for Genesys. The GVP IPCS is
installed behind a SIP server such that all interactions between the GVP IPCS and the SIP PBX go through
the SIP server.
IVR calls must be routed through a route point in order to support CTI monitoring.

Call Recording Flow

The diagram illustrates the flow for Genesys IVR call recording.
 

1 When a call arrives at an IVR Port DN, the SIP PBX server sends an INVITE to the Genesys SIP server.

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2 The Genesys SIP server forwards the INVITE to the destination IVR port on the Genesys GVP IPCS.
3 The Genesys GVP IPCS establishes the call and returns a 200 OK indication to the Genesys SIP server.
4 The Genesys SIP server sends a 200 OK indication to the SIP PBX to indicate that the call has been
answered.
5 The Genesys SIP server sends CTI events on the IVR port to the Recorder Integration Service for
recording and tagging.
6 The Recorder Integration Service requests that the IP Recorder record the call. The IP Recorder records
the call by:
a. Intercepting the RTP streams in the audio path.
b. Intercepting the SIP messages between the Genesys SIP server and the Genesys GVP IPCS to use
call information.
c. Correlating the CTI with the audio based on X-Genesys-CallUUID.

Configure the Recorder for IVR

Configure the switch to support SIP interception, as described in Configure the Recorder, page 1, using
the following settings:
1 In Recorder Manager, configure the IP Recorder for SIP Call Recording, with SIP protocol enabled.
2 In Enterprise Manager, configure a Phone data source, with a Switch/Sub Type of Genesys.
3 In SIP Trunk environments where the Genesys IVR is "front of switch," enable route point tracking to
access IVR ports and collect CTI data.
a. In the Genesys Phone data source, under Advanced Settings, click Add.
b. In the Key field, type TrackRoutePoints.
c. In the Value field, type true. When set to true, the Recorder Integration Service tracks route
point calls in the model.
d. Click Add.
e. In the Key field, type MarkRoutePoints.
f. In the Value field, type true. When set to true, the Recorder Integration Service creates
sessions/contacts for route point calls. (By default this setting is disabled and route point calls are
not recorded.)
g. Click Save.
4 Configure the Genesys SDK adapter to connect to the Genesys SIP server. Enable the Get RTP
Information setting.

Record IVR segments in Trunk Delivery

In Genesys trunk delivery recording of IVR (i.e., non-CTI based recording), there are two Media Processing
Component (MPC) parameters you must change in order to prevent excessive segmentation on single IVR
calls.
Configure the following parameters in Genesys Administrator, on the Provisioning > Environment >
Applications > [Media Control Platform] > Options tab, in the MPC section.

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Genesys SIP Server Integration Advanced Deployments

Parameter Set to

rtp.inputmode continuous
This setting specifies the input mode of
incoming RTP streams, and is set to vad by
default.

rtp.sendmode continuous
This setting specifies the output mode of
outgoing RTP streams, and is set to vad by
default.

Configure contact grouping

To implement contact grouping of IVR calls with call center calls, use the Contact Playback Gathering
feature in Cradle-To-Grave. In the Genesys environment, select Custom Data as the field by which to
group interactions. Then define a Custom Data attribute with ConnectionId as provided by Genesys.

Related information
"Configuring Cradle-To-Grave for Custom Data Mode" in the Interactions & Analytics Administration Guide

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Genesys SIP Server Integration Maintenance

Maintenance
Any changes to the extensions that are being recorded, or to the resources being used to record, may
necessitate changes to the data source used by the Genesys adapter.

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Genesys SIP Server Integration Troubleshooting

Troubleshooting
Failed to Connect to Genesys T-Server
Issue: This error message in the Genesys Adapter log indicates that the adapter is unable to
communicate with the Genesys T-Server.
Cause: Genesys Adapter improperly configured in the Recorder Manager. Define a Genesys Voice
Platform SDK Adapter if using Genesys T-Server version 8.0; define a Genesys (Legacy T-lib) Adapter if
using Genesys T-Server version 7.2 or earlier.
Resolution: Use Recorder Manager to view the Genesys adapter configuration information and verify
that the T-Server Hostname, T-Server Port, Application Name, and Application Password (if
required) match the information provided by the customer’s Genesys Server Administrator.
Cause: Network problem between the Integration Service and the Genesys T-Server.
Resolution: Verify network connectivity between the Integration Service and the Genesys T-Server using
the hostname and port information provided by the customer’s Genesys Server Administrator.
Cause: The Genesys T-Server is not presently functioning.
Resolution: Have the customer’s Genesys Server Administrator verify that the Genesys T-Server is
currently operational.

Troubleshooting

Failed to Connect to Genesys T-Server

Issue: This error message in the Genesys Adapter log indicates that the adapter is unable to
communicate with the Genesys T-Server.
Cause: Genesys Adapter improperly configured in the Recorder Manager. Define a Genesys Voice
Platform SDK Adapter if using Genesys T-Server version 8.0; define a Genesys (Legacy T-lib) Adapter if
using Genesys T-Server version 7.2 or earlier.
Resolution: Use Recorder Manager to view the Genesys adapter configuration information and verify
that the T-Server Hostname, T-Server Port, Application Name, and Application Password (if
required) match the information provided by the customer’s Genesys Server Administrator.
Cause: Network problem between the Integration Service and the Genesys T-Server.
Resolution: Verify network connectivity between the Integration Service and the Genesys T-Server using
the hostname and port information provided by the customer’s Genesys Server Administrator.
Cause: The Genesys T-Server is not presently functioning.
Resolution: Have the customer’s Genesys Server Administrator verify that the Genesys T-Server is
currently operational.

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Genesys SIP Server Integration Troubleshooting

Incorrect Call Tagging

Call Type (Direction) tagged as "Unknown"

Issue: The following issues stem from the same root cause:
• Call Type (Direction) tagged as "Unknown"
• Call Type (Direction) tagged incorrectly. Inbound calls tagged as outbound or outbound calls
tagged as inbound.
• Wrong extension number tagged; calls not recorded.
Cause: The WFO IP Capture engine requires that SIP Proxy IP Addresses be configured in
IPCaptureConfig.xml file to properly tag direction and extension for SIP Calls. If Proxy IP Addresses are
wrongly configured or not configured, the IP Recorder cannot determine the SIP Call direction correctly.
Resolution: Verify proxy configuration. If you make any changes to proxy configuration, you need to
restart the Recorder IPCaptureEngine Service for the changes to take effect. Note that if IP Capture is not
able to determine the direction, it will tag call type as "unknown" and raise an "UnknownCallDirection"
alarm with corrective action.

Incorrect Call Tagging

Call Type (Direction) tagged as "Unknown"
Issue: The following issues stem from the same root cause:
• Call Type (Direction) tagged as "Unknown"
• Call Type (Direction) tagged incorrectly. Inbound calls tagged as outbound or outbound calls
tagged as inbound.
• Wrong extension number tagged; calls not recorded.
Cause: The WFO IP Capture engine requires that SIP Proxy IP Addresses be configured in
IPCaptureConfig.xml file to properly tag direction and extension for SIP Calls. If Proxy IP Addresses are
wrongly configured or not configured, the IP Recorder cannot determine the SIP Call direction correctly.
Resolution: Verify proxy configuration. If you make any changes to proxy configuration, you need to
restart the Recorder IPCaptureEngine Service for the changes to take effect. Note that if IP Capture is not
able to determine the direction, it will tag call type as "unknown" and raise an "UnknownCallDirection"
alarm with corrective action.

Call Type (Direction) tagged incorrectly

See explanation above.

Wrong extension number tagged; calls not recorded

See explanation above.

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Genesys SIP Server Integration Troubleshooting

Failed to Record Calls in IP Interception Environment

Issue: Failed to record calls.
Cause: The extension is not present in the From/To headers.
Resolution: In some Genesys environments incoming calls contain an extension in the INVITE URI rather
than in the To header and outgoing calls contain an extension in the Contact header rather than in the
From header. To extract the extension from INVITE URI/Contact header, you need to set UseInviteURI
parameter in the IPCaptureConfig.xml file to true. For more details, see the Configure the UseInviteURI
section.

Failed to Record both parties on the call in Genesys Delivery

Issue: Failed to record calls for both parties on the call.
Cause: RIS receives two SIP INVITE from the Genesys switch. Each INVITE includes the other party on the
call, therefore RIS instructs the Recorder to record two inums that cannot be played on top of one
another.
Resolution: The IVR profile must be configured to record both parties on a call. See Configure the
Interactive Voice Response profile, page 28.

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 C h a p t e r 3  

Genesys Business Continuity

Integration

The Recorder integration with Genesys Business Continuity allows you to track calls across two
Genesys sites for calls delivered to a single Recorder Integration Service server, without the need
to configure the same extensions twice.

Topics 

Genesys Business Continuity Overview 56

Fallback behavior 57
Configure Genesys Business Continuity 58
Genesys Business Continuity Integration Genesys Business Continuity Overview

Genesys Business Continuity Overview

Genesys Business Continuity provides geographic redundancy for Genesys SIP servers deployed across
multiple sites. Pairs of SIP servers are distributed, but configured identically, with agent phones configured
to register against one or both sites, according to availability. If registered to both sites, calls can come
from either site.

In this solution, the Recorder Integration Service obtains CTI data and supports continuity by connecting
to both sides of the Genesys implementation simultaneously. Since agents may be registered to both
sites at once, a call to such an agent creates two independent calls with common characteristics. To
accommodate this, configuration on the Recorder side uses a parent-child data source model, with
settings such as extensions configured on the parent and then shared to the child data sources.

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Genesys Business Continuity Integration Fallback behavior

Fallback behavior
When all CTI links are up, the number of active links among all active Child Data Sources is reported back to
the Recorder Integration Service. Calls placed at/to either site within a pair of distributed Genesys SIP
Servers will be managed at the associated Child Data Source.
Mastership is negotiated between the Recorder Integration Services by counting the total number of
adapters for the Parent Data Source, just as in a standard environment. If any one CTI link goes down,
then all the extensions are placed into fallback at the Parent Data Source level. All calls at the Child Data
Source at that CTI link are closed, while calls on the alternate site will continue to be tracked on the Child
Data Source for that site.

 Deployment of two pairs of Recorder Integration Service servers as a means to achieve

redundancy is not supported. The Genesys Business Continuity system must deliver recordings
to a single Recorder Integration Service server.

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Genesys Business Continuity Integration Configure Genesys Business Continuity

Configure Genesys Business Continuity

To support Genesys Business Continuity you must create a "Parent" data source that reflects details
common to both SIP servers. You will then create an associated "Child" data source to represent each
specific site.

Before you begin 

Configure the Genesys SIP Server integration, as described in Configuration, page 27. When configuring
the Recorder, you will create a Phone data source with Genesys as the Switch/Sub Type. Within this data
source and its member group, configure all of the extensions you want to record, as well as any other
setting s that are common to both servers in your Genesys SIP pair (for example, extensions). This is your
Parent Data Source.
Once you have completed the configuration for your particular environment (VoIP Station-side
Interception, VoIP Station-side Delivery, or VoIP Gateway Recording), complete the following procedure.

Procedure 
1 Create an additional Genesys Phone Data Source for each site.
2 For each Data Source, under Data Source Parent, select the Parent Data Source you created
previously. This establishes the parent-child relationship between the two.
3 For each Child Data Source, create a CTI adapter as described in CTI Adapters, page 38.

Related information 
Recorder Configuration and Administration Guide

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 C h a p t e r 4  

Genesys Chat Integration

To capture Genesys Chat interactions via the Genesys Interaction Server, configure the
integration described in this chapter.

Topics 

Genesys Chat Integration Overview 60

Workflow: Configure the integration 61
Genesys Chat Integration Genesys Chat Integration Overview

Genesys Chat Integration Overview

Genesys Chat and Genesys Interaction Server are components of Genesys' omnichannel solution. This
integration allows you to bring Genesys Chat-based communications into the Verint system. The Verint
Text Capture component obtains and records interaction data. A full transcript of each interaction is
created, tagged with the relevant metadata, and associated with an employee. Chat interactions are then
available for search and review in other applications.

Related information 
Text Recording Configuration Guide

Security
Genesys supports HTTPS and Transport Layer Security (TLS) to secure chat sessions. This integration in
turn uses TLS for its connections with Genesys APIs, when TLS is enabled on the Genesys side.
 

Redundancy
If the primary Text Capture service is unavailable, a secondary Text Capture service will take over chat
processing. No chat interactions are lost as long as at least one of these nodes is up.

Related topics 
High Availability for Text, page 79
 

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Genesys Chat Integration Workflow: Configure the integration

Workflow: Configure the integration

Configure this integration by identifying text to be recorded, and creating a connection between Genesys
and the Recorder through an adapter for Text Capture.

Before you begin 

Install and configure the Recorder as part of your WFO setup.

Workflow 
1 Obtain Genesys Interaction Server information, page 61
Gather agent and network information from the Genesys environment. This information is required
to configure the agents as users in the system, and to allow the integration adapter to connect with
the Genesys Interaction Server and Universal Contact Server.
2 Configure Recorder roles, page 62
Assign roles to the Recorder node to give it the ability to request data from external sources and to
store the content it receives.
3 Create a Genesys Chat data source, page 62
Configure a data source that identifies the Genesys Interaction Server to Text Capture.
4 Add agents to the system, page 63
Add the agents for whom you want to capture chats and associate them with the data source.
5 Genesys Chat, page 64
Create an adapter to connect Text Capture to connect to the Genesys Interaction Server and
Universal Contact Server to receive events.

Obtain Genesys Interaction Server information

Gather agent and network information from the Genesys environment. This information is required to
configure the agents as users in the system, and to allow the integration adapter to connect with the
Genesys Interaction Server and Universal Contact Server.

Before you begin 

Configure Genesys Interaction Server, Universal Contact Server, and Genesys Chat. See the Genesys
documentation for instructions.

Procedure 
1 Obtain a list of agent logon IDs for the Genesys Interaction Server agents you want to record. You will
use these IDs to add the agents as Employees in the Verint system.
2 From the Genesys environment, obtain the following to include in the adapter configuration:
• Interaction Server Host name/IP address and Port number
• Universal Contact Server Host name/IP address and Port number

What to do next 
Configure Recorder roles, page 62

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Genesys Chat Integration Workflow: Configure the integration

Configure Recorder roles

Assign roles to the Recorder node to give it the ability to request data from external sources and to store
the content it receives.

Before you begin 

Install the Recorder as described in the WFO Installation Guide.

Procedure 
1 Configure the Text Capture node.
a. In Enterprise Manager, go to System Management > Enterprise > Settings.
b. Select the Text Capture node, then click Server Roles.
c. Ensure that the following roles are enabled:
• Content Server
• Recorder Ingestion Web Service
• Text Capture
d. Click Save.
e. Expand the server node under Installations and select the Text Capture role.
f. Click Save.
For deployments with redundancy, see High Availability for Text, page 79 for additional configuration.

Related information 
Recorder Configuration and Administration Guide

What to do next 
Create a Genesys Chat data source, page 62

Create a Genesys Chat data source

Configure a data source that identifies the Genesys Interaction Server to Text Capture.

Procedure 
1 In Enterprise Manager, click Recording Management.
2 Under Data Sources, click Settings.
3 Click Create Data Source.
4 Select Application as the Type, and Genesys Chat as the Switch Sub/Type.
5 Give the data source a Name that identifies it as the data source for Genesys Chat. The name must
not include the characters []<>"&! or ?.
6 Under Recorder Settings, select the Default Text Language.
7 Associate the data source with Text Capture by selecting the appropriate role under Associated Text
Capture Installations.
8 Click Save.

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Genesys Chat Integration Workflow: Configure the integration

9 Complete the remaining Recorder configuration by following the instructions in the chapter "Set up
individual Recorders" in the Recorder Configuration and Administration Guide.
While data sources typically have an associated member group that identifies specific recording targets,
this integration requires only that you add the agents to the system as Employees, as described in the
next topic.

What to do next 
Add agents to the system, page 63

Add agents to the system

Add the agents for whom you want to capture chats and associate them with the data source.

Procedure 
1 In Enterprise Manager, click User Management > Profiles.
2 Follow the procedure "Configuring employee profiles" in the WFO User Management Guide to add an
Employee Profile for an agent. When specifying the Agent ID, enter the ID for the agent in the field
beside the Genesys Chat data source you created previously. This ID should be the unique User Name
that the agent uses to log into the Genesys Workspace.

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Genesys Chat Integration Workflow: Configure the integration

3 Click Save.
4 Repeat for each agent whose chat interactions you want to capture.

What to do next 
Genesys Chat, page 64

Genesys Chat
Create an adapter to connect Text Capture to connect to the Genesys Interaction Server and Universal
Contact Server to receive events.

Procedure 
1 In Recorder Manager, navigate to General Setup > Integration Service > Settings.
2 Click Create, then select Genesys Chat Adapter.
3 Specify settings for this adapter in the right-hand pane, referring to the following table (all fields are
required unless otherwise indicated):

Field Description

Adapter Name Type a unique name for this adapter. Do not use any special
characters or characters that truncate XML. This field is not case
sensitive.

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Genesys Chat Integration Workflow: Configure the integration

Field Description

Description Optional: Type a description of the adapter.

Adapter Type A read-only field that lists the adapter selected above.

Startup Type Select a startup type:

• Automatic - Adapter starts automatically whenever the server is
started.
• Manual - Adapter starts only when a user clicks the Start button.
• Disabled - Adapter is disabled at startup, requiring that you edit
it to one of the above before it can start.

DataSource Select the Application data source that you created previously for
Genesys Chat.

Application Name A unique name for the application connection to Genesys.

Interaction Server Enter the host name or IP address of the Genesys Interaction Server
Host Host.

Interaction Server Enter the port number of the Genesys Interaction Server Host.
Post

Use Secondary Select this check box to enable failover if the primary Genesys
Interaction Server Interaction Server goes down.
for failover

Secondary Enter the host name or IP address of the backup Genesys Interaction
Interaction Server Server Host.
Hostname

Secondary Enter the port number of the backup Genesys Interaction Server
Interaction Server Host.
Port

Universal Contact Enter the host name or IP address of the Genesys Universal Contact
Server Host Server Host.

Universal Contact Enter the port number of the Genesys Universal Contact Server Host.
Server Port

Use Secondary Select this check box to enable failover if the primary Genesys
Universal Contact Universal Contact Server goes down.
Server for failover

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Genesys Chat Integration Workflow: Configure the integration

Field Description

Secondary Universal Enter the host name or IP address of the backup Genesys Universal
Contact Server Contact Server Host.
Hostname

Secondary Universal Enter the port number of the backup Genesys Universal Contact
Contact Server Port Server.

Security Settings Configure either a secure or non-secure connection. Select one of the
following:
• No Security- Creates a non-secure connection.
• TLS using Keystore - If you select this option, you must also
enter a Keystore Filename and a Keystore Password. Note that
the keystore password should be the same as your private key
password.
• TLS using Verint system SSL certificates - Uses Verint
system certificates from their default location.

4 Click Save.

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 C h a p t e r 5  

High Availability for Voice and

Screen

This section covers how to configure High Availability (also referred to as Redundancy) and
load balancing for voice and screen recording. It also covers modes of failover and disaster
recovery available on the Genesys side.

Topics 

N+N Redundancy 68
N+M Redundancy 69
Oracle SBC Redundancy 74
Oracle SBC Load Balancing 75
Genesys High Availability and Disaster Recovery 76
 
High Availability for Voice and Screen N+N Redundancy

N+N Redundancy
N+N redundancy is supported only with VoIP Interception solution in Genesys environment.
When configuring member groups in N+N (see Create and Edit Member Groups and Extensions in the
Recorder Configuration and Administration Guide), the member group settings you should choose for
Genesys are as follows:
 

Mode Primary Recorder Configuration

  Member Group Recorder Control Type

VoIP Interception IP Extension Pool Recorder or CTI Controlled

Note: If using a load balancer*, set a
load balancing type of Media Only or
Media with Signaling, and refer to the
specific guidelines set out in the
previous section.
If you require a higher channel count
than that of a single Recorder, use
Shared Interception as the load
balancing type.

 It is not necessary to set a load balancing type for this scenario.

* In this scenario you must ensure that each pair of N Recorders will receive identical signaling and media.
See Supported Environments in the N+N section of the Recorder Configuration and Administration
Guide for more details.

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High Availability for Voice and Screen N+M Redundancy

N+M Redundancy
An N+M system is configured with more recording resources than necessary to prevent audio or CTI loss
in any case of individual Recorder failure, as well as any network disconnection scenario. A primary
Integration Service paired with a secondary Integration Service work in concert to determine which of the
redundant M Recorders to task with recording. The two primary N+M scenarios are N+M All Shared and
N Dedicated + M Shared.
When configuring member groups in N+M (see Create and Edit Member Groups and Extensions in the
Recorder Configuration and Administration Guide), the member group settings you should choose for
Genesys are as follows:

Mode Environment Settings

N+M All Shared Genesys DMS • IP Extension Pool Member Group.

Delivery • Set the Recorder Control Type to Duplicate
Stream or External Controlled.

N+M All Shared with Genesys IP • IP Extension Pool Member Group.

Load-Balancing Interception • Set the Recorder Control Type to Recorder
Controlled or CTI Controlled.
• Set the Load-Balancing type to Media Only.

N+M All Shared with Genesys IP • IP Extension Pool Member Group.

Shared Interception Interception • Set the Recorder Control Type to Recorder
(no Load-Balancing). Controlled or CTI Controlled.
• Set the Load-Balancing type to Shared
Interception.

 While the following diagrams depict scenarios involving two Integration Services, N+M all shared
or N dedicated + M shared can be accomplished using only a single Integration Service.

N+M All Shared Scenario

N+M All Shared involves a pool of M Recorders all configured in the same way, with load balancing—via
the Integration Service or a 3rd party device—to distribute calls to Recorders. N+M All Shared scenario is
supported with the Genesys VoIP delivery solution, as well as with the VoIP interception solution.

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High Availability for Voice and Screen N+M Redundancy

Call Center Call Center

CTI Server CTI Server
/Voice /Voice
Switch Switch

Integration
Integration
Service
Service
Server
Server

Recorder
Recorder
Recorder 1 Recorder 2 N+M
N +1
Recorder N

N+M All Shared

1+1 Integration Service Redundancy with VoIP Delivery

It is possible to configure a pair of redundant Integration Services - primary and secondary (1+1
Integration Service Redundancy). Configure both Integration Service addresses on the Genesys side, to
enable the GVP Resource Manager to initiate a SIP dialog with either of them. Configure GVP Resource
Manager to contact the Primary Integration Service first, and to fall back to the Secondary Integration
Service only if the primary is not responsive.

N Dedicated + M Shared Scenario

In N Dedicated + M Shared, N Recorders are paired with shared M Recorders. These Recorders remain
in standby mode until an N Recorder fails, at which time an Integration Service server will instruct an M
Recorder to take over on that channel or extension. This scenario is not supported with Genesys VoIP
delivery.
 

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High Availability for Voice and Screen N+M Redundancy

Call Center
CTI Server
/Voice
Switch

Integration
Integration
Service
Service
Server
Server

Recorder
Recorder N Recorder
M
Recorder 1 Recorder 2 1

Recording System with N Redundant Recording System

Dedicated Recorders with M Shared Recorders

N Dedicated + M Shared
In environments that are N dedicated trunk side (either TDM or SIP trunk) and M shared (all types), if the
dedicated trunk side Recorder experiences issues (indicated by one or more alarms on the Recorder
Manager status page for that Recorder), the Integration service will engage the shared Recorder to record
a call. The dedicated Recorder may still be able to record a call, even though a problem was reported.
Therefore, it is possible that the system will have a duplicate recording of a call, and two sessions will
appear in the portal as having the same attributes and audio. This is expected behavior.
For example, a dedicated SIP trunk Recorder may discover a loss of packets for some calls that occurred
in the past. An alarm is raised indicating a minor issue with the Recorder. For future calls, Integration
Service will engage a shared Recorder to record the calls, in addition to the dedicated Recorder that may
or may not lose some packets. While the problem with the dedicated Recorder exists, duplicate sessions
may be found for each call in the portal—one recorded by the dedicated Recorder (and which may have
inferior audio quality) and another one recorded by the shared Recorder (with good audio quality).

Real-time/Record-on-Demand
Redundancy and real-time/record-on-demand (ROD) are supported as long as the Integration Service is
controlling an active recorder, and a real-time audio or ROD session can be initiated.

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High Availability for Voice and Screen N+M Redundancy

Using Corero and/or Link Protectors in an N+M All Shared Environment

You can configure redundant Corero devices in an N+M all-shared environment.

Recorder 1 Recorder 2 Recorder 3

Recorder 1

Corero Device

To create this type of setup, do the following:

• Create a Monitor Group on the Corero device for the Recorder, in order to load balance RTP
streams across the recorders in the group.
• Use Call Control Flooding to distribute the Call Control Protocol to all the required systems. This
will ensure that all Recorders see the same Call Control Packets.
• When using the Corero device, set all 100 MB ports explicitly to 100 MB full duplex and not auto-
sense. Monitor Groups must not be configured to ignore link status, as this will result in the device
not being able to detect a recorder network failure.
The Recorder also supports the use of standard link protection hardware. When selecting a link protector,
it is recommended that the model selected has support for power failure fail-through. The link protector
should typically be used to build 1+1 warm standby solutions. This warm standby could be for either the
Recorder or the Corero device. Where several links into a device must be protected, consideration should
be given to scenarios involving a single link failure, to ensure recording would continue in accordance with
the design.

SIP Recording Configuration

For SIP recording, please note that you must complete following additional configuration steps.
1 Add UDP SIP as a new application on port 5060 and protocol UDP, and enable flooding configuration
for this application. This ensures that all SIP messages are flooded to all recorders.

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High Availability for Voice and Screen N+M Redundancy

2 Exclude the UDP SIP application from load balancing in the Monitor Group input. The Monitor Group
input processes all UDP packets. As UDP SIP is already being flooded through the flooding
configuration, it doesn't require additional load balancing to the recorders. Failing to exclude UDP SIP
will result in duplicate UDP SIP messages to the Recorder to which this traffic is routed; one through
load balancing and the other through flooding.
See “Configure SIP Interception” in the Recorder Configuration and Administration Guide for general
configuration instructions.

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High Availability for Voice and Screen Oracle SBC Redundancy

Oracle SBC Redundancy

This Oracle SBC solution provides redundancy on the SBC side, and an N+1 solution on the Verint
Recorder side. Should a Recorder fail, the Oracle SBC can be configured to automatically replicate the
active session to a backup Recorder. This can be accomplished via a configurable SBC SIP heartbeat
feature.
N+1 on the Recorder means that there are “N” servers with a set of channels dedicated to recording, “+1”
server dynamically allocated as a backup should any part of group N become unavailable.

 You must configure the SBC Call Recording Server (CRS) object for the "+1" Recorder as the
Secondary Address for the "N" CRS objects.

 SBC can only send traffic to one Recorder at a time.

Please see the VoIP Delivery, VoIP Interception without Top Layer, or VoIP Interception with Top Layer
Deployment Reference Guide for redundancy information specific to your recording environment.

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High Availability for Voice and Screen Oracle SBC Load Balancing

Oracle SBC Load Balancing

The capacity of Oracle SBC can reach tens of thousands of sessions. This is more concurrent calls than an
individual Verint WFO IP Recorder can process. For this reason you must configure Oracle SBC such that a
group of Recorders receives even load distribution.
To set up SRR Load Balancing, complete the following tasks, all on the Oracle SBC side (please refer to the
Oracle SBC documentation for detailed instructions):
• Create one Session Agent (each associated with a realm) for every Recorder. Each Session Agent
must point to the SIP Trunk IP address on the switch.
• Assign a Session Agent Group (SAG) to the Session Agents associated with the Recorders/call
recording realms are assigned to a SAG.
• Enable Load Balancing for the SAG. Set one of the following Load Balancing options:
• Least Busy—recommended to evenly distribute the load when all Recorders have the same
capacity.
• Proportional—recommended when the Recorders have different capacities.
• Set up a Realm for each Recorder. The Realm definition includes a SIP Interface (that is, the IP
address where the SIP Trunk terminates on the SBC).
• Create a CRS for each Recorder, and assign the CRSs to a corresponding Realm. The CRS defines
the IP address of the Recorder server, identifying for the SBC where it should send the replicated
SIP/RTP traffic.
As the SBC load balances new calls across these Realms, the call recording servers will get the replicated
packets in a load-balanced manner.
To add a third primary CRS to this configuration for inbound calls, add another CRS object and set its
secondary address as the as that of the server used for failover. Add a realm for this CRS (including its
associated SIP Interface and Session Agent). Add the Session Agent of the new CRS to the SAG, in order to
include this CRS in the Load Balancing group.

 For recording both inbound and outbound calls, you should replicate the inbound configuration
described above on the outbound side of the SBC (that is, the side that terminates the SIP Trunks
from the service provider). Note that in order to provide load balancing for outbound calls,
multiple SIP Trunks to the service provider are required, one for each CRS.

 If you configure Load Balancing but do not configure Redundancy, a failed Recorder will result in
the loss of recordings (that is, calls will not be redistributed across active reocrders). To ensure
that all calls are recorded as desired, configure Redundancy in addition to Load Balancing.

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High Availability for Voice and Screen Genesys High Availability and Disaster Recovery

Genesys High Availability and Disaster Recovery

There are a couple of options on the Genesys side that you implement to make operations more robust:
• server warm/hot standby features (for general resiliency)
• SIP Business Continuity (specifically for disaster recovery)

 Genesys top-down or round robin load balancing is also supported, as long as only one SIP dialog
is created for the call.

Genesys T-Server Hot and Warm Standby

The Recorder supports both Hot Standby and Warm Standby deployments of the Genesys Server, but
the behavior for each one will be different:
• In a Hot Standby deployment, if the primary Genesys Server goes down the move to the
secondary Genesys Server is seamless and recording will continue without any impact.
• In a Warm Standby deployment, if the primary Genesys Server goes down, the Recorder Manager
relies on the configuration within the Genesys SDK adapter to configure the primary and
secondary servers. Once the primary fails and the connection is lost, the Recorder will attempt to
connect to the secondary. All recordings in progress lose tagging during this process, so there is
some interruption in recording functionality.

SIP Business Continuity Deployment

Genesys SIP Server 8.1 and higher supports SIP Business Continuity, an architecture that allows for
geographic redundancy, also referred to as Disaster Recovery (DR). The solution involves paired sites
(enterprise redundancy) and may include High Availability Severs (by means of warm/hot standby as
described above) within each site.
In geographic redundancy, deployments are dispersed across two physical locations, with a server (or pair
of servers) situated in each. In these environments, two Genesys T-Servers are active (as opposed to
being in a hot standby configuration) and are configured with all extensions across both sites.
Agent SIP Phones are typically configured to maintain simultaneous registrations with both sites’ SIP
Servers, whereas Agent Desktop applications maintain a login to only one peer at any given time. When
an agent logs in, the desktop, by default, connects and logs into the SIP Server at the agent's preferred
site, and that site provides calls and CTI events for that agent's calls. If the desktop is unable to log in
initially to the SIP Server at the preferred site, it switches over its connection to the other site. These
switch overs are referred to as DR switch overs. Under normal conditions, only the SIP server at the
preferred site will provide calls and CTI events. Note that since the phones are simultaneously registered
at both sites, successful CTI registration on the non-preferred site does not mean that this SIP server
provides calls and CTI to the phone — this will only happen in the event of a failure.

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High Availability for Voice and Screen Genesys High Availability and Disaster Recovery

Both Interception and Delivery solutions should have enough Recorder capacity at each site for the full
population of agents from both locations.

VoIP Station-side Interception

In Interception environments, the deployment should consist of either a single Recorder Integration
Service or a Recorder Integration Service 1+1 pair, plus two groups of Recorders (one group per site).
If the customer wants to maintain recording during Disaster Recovery failures, the Recorder Integration
Service and Recorder deployment design should take into account that the full agent population may
need to be recorded from each site, in case a Disaster Recovery failover occurs. This will impact system
sizing, and potentially create a requirement for load balancing.
To configure the Recorder for VoIP Interception, complete the following tasks (referring to the Recorder
Configuration and Administration Guide for step-by-step instructions).
1 Create two data sources, one for each site. For each, configure a Switch Signaling Interface IP
Address to allow the Recorder identify the data source each call is meant for.
2 Within these data sources, member group configuration depends on where interception occurs, and
on the load balancing type.
• If using station side interception with no load balancing, or load balancing of the type Media
Only, or Media with Signalling, it is sufficient to create a single member group in each data
source. This member group will contain all Recorders and extensions from across sites.
• If using station side interception with a load balancing type of Shared Interception, each data
source should contain a member group for each site, configured with the extensions particular to
that site.

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High Availability for Voice and Screen Genesys High Availability and Disaster Recovery

• If intercepting the station side traffic at the gateway (referred to as gateway infrastructure
interception), then a single member group in each data source containing all Recorders and
extensions will suffice. In addition, IP Analyzers would be used in this scenario, to insure that
Recorders receive signaling and audio for the calls it needs to record.
3 Create Recorders at each site, all grouped, with each Recorder assigned to both data sources.
If Interception is occurring at the gateway, it is possible that each gateway can deliver calls to either
Genesys SIP server in a pair. In this case, you must have IP analyzers to forward signalling across to the
other location.

VoIP Station-side Delivery (SIP Trunk with Oracle SBC or SIP Trunk Interception)

 For Oracle SBC with SIP trunk, and SIP Trunk Interception recording solutions, each Genesys
switch requires dedicated Recorders at each site.

Since each switch will have dedicated Recorders, the Oracle SBC must also be configured so that signalling
and audio is presented to the specific Recorders responsible for recording the switch that handles the call.
You can accomplish this by configuring recording realms on the SBC, based on the Genesys SIP server for
which each call is destined.
To configure the Recorder for VoIP Delivery, complete the following tasks (referring to the Recorder
Configuration and Administration Guide for step-by-step instructions).
1 Create two data sources, one for each site.
2 Configure these data sources with all extensions included with the standard station side delivery
configuration you configured under VoIP Station-side Delivery, page 32.

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 C h a p t e r 6  

High Availability for Text

This section covers how to configure High Availability (also referred to as Redundancy) and for
Text Capture integrations.

Topics 

Overview 80
Configure High Availability for Text 81
High Availability for Text Overview

Overview
Text capture uses a 1+1 redundancy model. You may pair a Primary Text Capture node with a secondary
Text Capture node using secondary Role associations in Enterprise Manager. The primary will be have initial
Primary control over the secondary.
In a deployment with Text Redundancy, when the primary Text Capture service is unavailable, the
secondary Text Capture service picks up chat processing. When the primary comes back up, it takes back
over. No chat interactions are lost as long as one of the nodes is up.
 

Genesys Server Warm Standby

The Recorder supports Warm Standby deployments for Genesys Interaction and Universal Contact
Servers. If the primary Genesys Server goes down, the Recorder Manager relies on the configuration
within the Genesys adapter to configure the primary and secondary servers. If the primary fails and the
connection is lost, the Recorder will attempt to connect to the secondary. All recordings in progress lose
tagging during this process, so there is some interruption in recording functionality.

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High Availability for Text Configure High Availability for Text

Configure High Availability for Text

 
For deployments with redundancy, pair the primary Text Capture node with a secondary Text Capture
node, allowing the secondary to act as a backup.

Procedure 
1 Configure two Text Capture nodes as follows:
a. In Enterprise Manager, go to System Management > Enterprise > Settings.
b. Select the Text Capture node, then click Server Roles.
c. Ensure that the following roles are enabled:
• Content Server
• Recorder Ingestion Web Service
• Text Capture
d. Click Save.
e. Expand the server node under Installations and select the Text Capture role.
f. Click Save.
2 Select a Text Capture role under the primary node from step 1.
3 Click the Secondary Role screen.
4 Under Paired Secondary Role, select the Text Capture role under the other node created in step 1.
5 Click Save.

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 C h a p t e r 7  

Set Up Certificates

This chapter describes how to set up certificates so that you may secure communications using
TLS/SSL for the Recorder Integration Service, and for the Text Capture adapter in Genesys Chat
environments.

Topics 

Overview 83
Certificates for Genesys Platform SDK and Interaction Server 84
Certificates for SBC 90
Set Up Certificates Overview

Overview
You can secure communications between Verint and the Genesys Platform SDK, a Genesys Interaction
Server, or an SBC. To do so, you must set up certificates for the components involved. See:
• Certificates for Genesys Platform SDK and Interaction Server, page 86
• Certificates for SBC, page 90

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Set Up Certificates Certificates for Genesys Platform SDK and Interaction Server

Certificates for Genesys Platform SDK and

Interaction Server
When the Genesys T-Server is configured in secure mode, you should configure Verint's Genesys SDK
adapter to use a TLS connection. This will allow encrypted CTI event messages to be transported from the
Genesys Platform SDK or Genesys Interaction Server to the Verint platform, where they will be
decrypted.
There are a couple of different ways to configure certificates and authentication, depending on your
setup.

Certificates
Configuration using a custom JKS file
Create a new JKS key store file to hold the client certificate, which will be used for the key manager for
validation.
Specify the path for the keystore and its password when you configure the adapter. The password for the
JKS file and the client private key password must be the same.
See: "Create new keys using a self-signed certificate" or "Create new keys using a CA-signed certificate" in
Set up keys for certificates, page 90.

Configuration using Verint default certificates

You can use Verint’s existing server certificate as client certificate for secured communication with
Genesys.
See: "Use existing keys" in Set up keys for certificates, page 90.

Authentication
Whether you need to provide a certificate to Genesys for authentication depends on whether you are
using 'mutual' authentication on the Genesys side.

Mutual authentication
If the Genesys server requires mutual authentication you must provide it with a certificate that is trusted
by Genesys.
This can be either included in the custom JKS or, if SSL is configured on the server, using Verint standard
client certificates from IMPACT360/Conf/Security/cert.pem.
See Provide a certificate to Genesys, page 88.

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Set Up Certificates Certificates for Genesys Platform SDK and Interaction Server

Non-mutual authentication
If you are not using mutual authentication, skip the tasks under Provide a certificate to Genesys, page 88.

Workflow
To set up TLS for Genesys, do the following:
• Set up keys for certificates, page 90
• If you are using mutual authentication on the Genesys side, Provide a certificate to Genesys,
page 88.
• Configure the adapter for TLS, page 89

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Set Up Certificates Certificates for Genesys Platform SDK and Interaction Server

Certificates for Genesys Platform SDK and Interaction Server

When the Genesys T-Server is configured in secure mode, you should configure Verint's Genesys SDK
adapter to use a TLS connection. This will allow encrypted CTI event messages to be transported from the
Genesys Platform SDK or Genesys Interaction Server to the Verint platform, where they will be
decrypted.
There are a couple of different ways to configure certificates and authentication, depending on your
setup.

Certificates
Configuration using a custom JKS file
Create a new JKS key store file to hold the client certificate, which will be used for the key manager for
validation.
Specify the path for the keystore and its password when you configure the adapter. The password for the
JKS file and the client private key password must be the same.
See: "Create new keys using a self-signed certificate" or "Create new keys using a CA-signed certificate" in
Set up keys for certificates, page 90.

Configuration using Verint default certificates

You can use Verint’s existing server certificate as client certificate for secured communication with
Genesys.
See: "Use existing keys" in Set up keys for certificates, page 90.

Authentication
Whether you need to provide a certificate to Genesys for authentication depends on whether you are
using 'mutual' authentication on the Genesys side.

Mutual authentication
If the Genesys server requires mutual authentication you must provide it with a certificate that is trusted
by Genesys.
This can be either included in the custom JKS or, if SSL is configured on the server, using Verint standard
client certificates from IMPACT360/Conf/Security/cert.pem.
See Provide a certificate to Genesys, page 88.

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Set Up Certificates Certificates for Genesys Platform SDK and Interaction Server

Non-mutual authentication
If you are not using mutual authentication, skip the tasks under Provide a certificate to Genesys, page 88.

Workflow
To set up TLS for Genesys, do the following:
• Set up keys for certificates, page 90
• If you are using mutual authentication on the Genesys side, Provide a certificate to Genesys,
page 88.
• Configure the adapter for TLS, page 89

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Provide a certificate to Genesys

The following is only required if you have configured mutual authentication on the Genesys side.

Provide a certificate for mutual authentication

When the Genesys T-Server is configured with the 'mutual=1' property, provide a .pem or .cert certificate
file to Genesys in one of two ways:
• Export a certificate from the JKS file if you have used that option. (See "Create new keys using a
self-signed certificate" or "Create new keys using a CA-signed certificate" in the previous section.)
• Provide the existing file from Impact360\Software\Conf\Security folder.
Import one of these client certificates to the Genesys server as a trusted certificate.

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Set Up Certificates Certificates for Genesys Platform SDK and Interaction Server

Configure the adapter for TLS

Configure the security settings for the Create a Genesys Voice Platform SDK Adapter, page 39 for TLS.
Note that the adapter must connect to the secure port (rather than the regular port) if TLS is enabled. The
specific port number will depend on your environment's configuration.

 Even when mutual authentication is not required, or you have not chosen the JKS option
described in the previous section, you must provide a path path and password to a dummy or
default JKS file below.

Configure the adapter

1 In Recorder Manager, click General Setup > Integration Adapters > Settings.
2 Select the Genesys SDK Adapter.
3 Under Security, select one of the following:
• No Security - Default existing behavior.
• TLS using key store - If you select this option, you must also enter a Key Store Filename, a
Keystore Password, and specify a Keystore Format (JKS or PEM). Note that the Keystore password
should be the same as your private key password.
• TLS using Verint System Keys - Select this option if you are using existing Verint keys.

4 Click Save.

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Set Up Certificates Certificates for SBC

Certificates for SBC

Use the following procedure to set up certificates to secure communication between and SBC and the
Recorder Integration Service. The SBC certificate and certificates used on the Verint side must both be
signed by the same Certificate Authority (CA). Do not use self-signed certificates or Verint-signed
certificates for this implementation.

 You may use the same certificate(s) for each Integration Service in a multiple Integration Service
deployment, or you can generate different ones for each.

Set up keys for certificates

Set up keys for both the SBC and the Recorder using CA-signed certificates. There should already be a
certificate installed on the SBC for itself. The commands in the tasks below are examples and are meant to
guide an end user in how to set up a keystore to secure the connection between the SBC and the Recorder.
There are alternative utilities and tool chains available that an end user may choose instead of executing
these steps verbatim.

 The *.jks files *.pem files in the following procedures are copied to the location of the keystore
(typically, “%IMPACT360SOFTWAREDIR%\conf”). A user may place the keystore anywhere on the
file system.

The following procedures use the file name 'Keystore.jks' as an example, you may name this file whatever
you like.

 When creating keystore passwords, choose a strong password, ideally using a combination of
letters and numbers, at least eight characters in length.

Create new keys using a Certificate Authority (CA)-signed certificate

If you do not have any certificates created for the server, or you want dedicated certificates for this
integration, then the following steps will guide you to create a new set of certificates.
1 Create a keystore:
a. Open the Windows command prompt, and change directories to the location of the keystore tool
(typically, “%IMPACT360SOFTWAREDIR%\conf”).
b. Enter the following:
%I360_JAVA_HOME%\bin\keytool.exe
-genkeypair -keyalg RSA -alias mykey
–validity 3650
-keystore Keystore.jks
The term “validity” defines how long the certificate will be valid before expiring, and is set in days.
In the example above, “3650” indicates that the certificate will expire after ten years. You can set
this value to any duration, but if you don’t include it the certificate will expire after 90 days.

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Set Up Certificates Certificates for SBC

 Enter the term “mykey” verbatim (that is, it is not meant to be replaced with text representing
your key).

c. Enter the keystore password (the password will be obscured).

d. When prompted, answer questions to generate a distinguished name (DN). When asked for the
first and last name that will be used as the common name (CN) within the DN, it is recommended
that your entry not contain any spaces, as this will cause problems with some third-party
providers.
e. Verify that the information in the distinguished name is correct when prompted. Make a note of
the CN for use in the following procedures.
f. Do not enter a new password for the generated key. The password must be the same as the
keystore.
2 Create a certificate a CA-signed public certificate for the Integration Service.
a. Export your certificate signing request for the CA to sign. In the command prompt, change
directories to the location of the keystore (typically, “%IMPACT360SOFTWAREDIR%\conf”), and
enter the following:
%I360_JAVA_HOME%\bin\keytool.exe -certreq -alias mykey -sigalg SHA1withRSA -file
SIPRIS.csr -keystore Keystore.jks
The alias must be the same one you used on step 1 (in this example, “mykey”). This command
creates a Certificate Signing Request (CSR) file. This is the file you will send to be signed.
b. Send the CSR file to the CA to be signed.
The CA will provide back a signed private key PEM file (SIPRIS.pem in this example) and the CA-
signed public key Privacy Enhanced Mail (PEM) file (CA_CERT.pem in this example).
3 Import the CA’s public key into the keystore. In the command prompt, change directories to the
location of the keystore (typically, ““%IMPACT360SOFTWAREDIR%\conf”), and enter the following
command:
%I360_JAVA_HOME%\bin\keytool.exe –import –trustcacerts –alias CACER -file CA_CERT.pem
-keystore Keystore.jks

 The alias used the represent the CA public key in this step must be the alias used when creating
the CA public key. If the CA public key is already in the trusted certificates for Windows, a
message indicating this will appear. Select Yes to add it into your keystore anyway.

4 Import the CA-signed key into the keystore. In the command prompt, change directories to the
location of the keystore (typically, “E:\Impact360\Software\conf\”), and enter the following
command:
%I360_JAVA_HOME%\bin\keytool.exe –import –trustcacerts –alias mykey –file SIPRIS.pem –
keystore Keystore.jks
The alias must be the same one you used on step 1 (in this example, “mykey”).

Import existing HTTPS certificates into a new keystore

If you already have certificates for the server or this integration and want to create a new keystore
containing those keys, these steps will guide you. This sequence is for importing certificates used to deploy
HTTPS encryption on the WFO suite, but can be adjusted as-needed to import another set of certificates.
1 Create a keystore:

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Set Up Certificates Certificates for SBC

a. Open the Windows command prompt, change directories to the location of the keystore tool
(typically, “%IMPACT360SOFTWAREDIR%\conf”) and enter the following:
%I360_JAVA_HOME%\bin\keytool.exe -importkeystore -srckeystore
%IMPACT360SOFTWAREDIR%\Conf\Security\svr_cert_key.p12 -srcstoretype PKCS12 -
destkeystore Keystore.jks
This command will import the encrypted public-private certificate information used to encrypt the
HTTPS traffic to the server into a new keystore to encrypt the SBC traffic. You will be prompted to
create a password for the new keystore and for the password for the encrypted PKCS12 container
with the HTTPS certificates. Take note of the alias of the imported certificate.
2 Change the private certificate alias:
a. Open the Windows command prompt, change directories to the location of the keystore tool
(typically, “%IMPACT360SOFTWAREDIR%\conf”) and enter the following:
C:\Impact360\Software\Conf>%I360_JAVA_HOME%\bin\keytool.exe -changealias –alias <noted
alias from 1> -destalias mykey -keystore Keystore.jks
This command will change the alias entry for the imported certificates to the standard ‘mykey’
format used in this guide.

Import the SBC certificates into keystore

Once you have a keystore housing the private and public certificate for the Recorder, you will need to
import the certificates representing the SBC.
1 Import the SBC CA certificate:
a. If the SBC is using a different CA than the Recorder, then you must import the SBC’s CA certificate
into the keystore as a trusted certificate. If the Recorder and SBC share the same CA, you can skip
this step.
b. Open the Windows command prompt, and change directories to the location of the keystore tool
(typically, “%IMPACT360SOFTWAREDIR%\conf”).
c. Enter the following:
C:\Impact360\Software\Conf>%I360_JAVA_HOME%\bin\keytool.exe -importcert -file <Root CA
cert> -alias SBCCA -keystore Keystore.jks
This command will import the root CA certificate chain within the keystore.
2 Import the SBC certificate:
a. Open the Windows command prompt, change directories to the location of the keystore tool
(typically, “%IMPACT360SOFTWAREDIR%\conf”) and enter the following:
C:\Impact360\Software\Conf>%I360_JAVA_HOME%\bin\keytool.exe -importcert -file <the_
cert> -alias CERTALIAS -keystore Keystore.jks
This command will import the the SBC certificate chain within the keystore.

Import Recorder certificates into the SBC

This task is optional. If you are using the same CA for both the the SBC and Recorder, the CA public
certificate is already trusted by the the SBC and this step can be skipped.
In order for the the SBC to communicate with the Recorder, the the SBC must trust the certificate supplied
by the Recorder.
1 Import one of the following to apply trust:

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Set Up Certificates Certificates for SBC

• the CA public certificate from Step 3 of "Create new keys using a Certificate Authority (CA)-signed
certificate."
or
• the specific CA-signed public certificate of the Recorder in Step 4 of "Create new keys using a
Certificate Authority (CA)-signed certificate."
If the previously imported public certificate is not available, export the signed certificate and provide it
for import to the the SBC.
Example:  
%I360_JAVA_HOME%\bin\keytool.exe -exportcert -alias mykey -keystore Keystore.jks -rfc
-file exportedcertificateforthe SBC.pem
Please review the the SBC documentation for more information on how to trust the Recorder certificate.

Use existing keys

You may use existing Verint keys for your certificates. If you have set up your Verint system as a whole to
use SSL (as described in the WFO Security Configuration Guide), you will already have established the files
you need, in the following locations:
• "IMPACT360/Conf/Security/cert.pem" – holds the Recorder’s server certificate.
• "IMPACT360/Conf/Security/key.pem" – holds the Certificate’s private key.
These files are as part of the general server SSL installation/configuration and are used as the server's
client certificate for SSL communication with Enterprise Manager and other components.
As such, if the the Recorder Integration Server is already configured for SSL, it may be a convenient option
for to provide its certificate (cert.pem) to external SIP end points for authentication of the certificate.

Provide your own keys

If the system is not already configured for SSL, use one of the other options to provide your own JKS file
(see Create new keys using a self-signed certificate, page 93 or Create new keys using a CA-signed
certificate, page 94).
Note that the Verint software installs some default keystore files (referenced by default in the adapter).
You may use these files if you not using mutual authentication. If you are using mutual authentication you
must to create a certificate, then create a JKS file and put the certificate inside it as described in Create new
keys using a CA-signed certificate, page 94.

 The following procedures use the file name 'SIPKeystore.jks' as an example, but you may name
this file whatever you like.

Create new keys using a self-signed certificate

1 Create a keystore:
a. Open the Windows command prompt, change directories to the location of the keystore tool
(typically, “E:\Impact360\Software\conf\”) and enter the following:

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E:\Impact360\Software\JDK\jre\bin\keytool.exe -genkeypair -keyalg

RSA -alias mykey –validity 3650 -keystore SIPKeystore.jks
The term “validity” defines how long the certificate will be valid before expiring, and is set in days.
In the example above, “3650” indicates that the certificate will expire after ten years. You can set
this value to any duration, but if you don’t include it the certificate will expire after 90 days.

 Enter the term “mykey” verbatim (that is, it is not meant to be replaced with text
representing your key).
b. Enter the keystore password (the password will not be obscured).

 Choose a strong password, ideally using a combination of letters and numbers, at least eight
characters in length.

c. Answer questions when prompted, in order to generate a distinguished name. When asked for the
first and last name that will be used as the common name (CN) within the distinguished name, it is
recommended that your entry not contain any spaces, as this will cause problems with some
third-party providers.
d. Verify that the information in the distinguished name is correct when prompted. Make a note of
the CN for use in the following procedures.
e. Do not enter a new password for the generated key. The password must be the same as the
keystore.
2 Create a self-signed public certificate for the Integration Service.
a. In the Command Prompt, change directories to the location of the keystore tool (typically,
“E:\Impact360\Software\conf\”), and enter the following:
E:\Impact360\Software\JDK\jre\bin\keytool.exe -exportcert -file
SIPKeystore.pem -keystore SIPKeystore.jks
b. Enter the keystore password when prompted.
3 A message indicating the location in which file the is stored will appear. It is normally in the local
directory. Make a note of this for use in the following procedures.

Create new keys using a CA-signed certificate

1 Create a keystore:
a. Open the Windows command prompt, change directories to the location of the keystore tool
(typically, “E:\Impact360\Software\conf\”) and enter the following:
E:\Impact360\Software\JDK\jre\bin\keytool.exe -genkeypair -keyalg
RSA -alias mykey –validity 3650 -keystore SIPKeystore.jks
The term “validity” defines how long the certificate will be valid before expiring, and is set in days.
In the example above, “3650” indicates that the certificate will expire after ten years. You can set
this value to any duration, but if you don’t include it the certificate will expire after 90 days.

 Enter the term “mykey” verbatim (that is, it is not meant to be replaced with text
representing your key).
b. Enter the keystore password (the password will not be obscured).

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Set Up Certificates Certificates for SBC

 Choose a strong password, ideally using a combination of letters and numbers, at least eight
characters in length.

c. Answer questions when prompted, in order to generate a distinguished name. When asked for the
first and last name that will be used as the common name (CN) within the distinguished name, it is
recommended that your entry not contain any spaces, as this will cause problems with some
third-party providers.
d. Verify that the information in the distinguished name is correct when prompted. Make a note of
the CN for use in the following procedures.
e. Do not enter a new password for the generated key. The password must be the same as the
keystore.
2 Create a certificate authority (CA)-signed public certificate for the Integration Service.
a. Export your certificate signing request for the CA to sign. In the command prompt, change
directories to the location of the keystore (typically, “E:\Impact360\Software\conf\”), and
enter the following:
E:\Impact360\Software\JDK\jre\bin\keytool.exe -certreq -alias
mykey -file SIPRIS.csr -keystore SIPkeystore.jks

 The alias must be the same one you used on step 1 (in this example, “mykey”). This command
creates a Certificate Signing Request (CSR) file. This is the file you will send to be signed.

b. Send the CSR file to the CA to be signed.

 The CA should sign the certificate using the same template used on CUCM. The CA will provide
back a signed private key PEM file (SIPRIS.pem in this example) and the CA signed public key PEM
file (VMS-PROD.pem in this example).

3 Import the CA-signed key into the keystore:

a. In the command prompt, change directories to the location of the keystore (typically,
“E:\Impact360\Software\conf\”), and enter the following command:
E:\Impact360\Software\JDK\jre\bin\keytool.exe –import –
trustcacerts –alias mykey –file SIPRIS.pem –keystore
SIPkeystore.jks

 The alias must be the same one you used on step 1 (in this example, “mykey”).

4 Import the CA’s public key into the keystore.

a. In the command prompt, change directories to the location of the keystore (typically,
“E:\Impact360\Software\conf\”), and enter the following command:
E:\Impact360\Software\JDK\jre\bin\keytool.exe –import –
trustcacerts –alias CACER -file CA_CERT.pem -keystore
SIPkeystore.jks

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 The alias used the represent the CA public key in this step must be the alias used when creating
the CA public key. The CA public key is already in the trusted certificates for Windows and a
message indicating this will appear. Select Yes to add it into your keystore anyway (as in the
image below).

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Verint Global Headquarters
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Suite 100
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[email protected]
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www.verint.com

© 2019 Verint Systems Inc.

All Rights Reserved Worldwide.
Confidential and Proprietary Information of Verint Systems Inc.

The contents of this material are confidential and proprietary to Verint Systems Inc.
and may not be reproduced, published, or disclosed to others without express
authorization of Verint Systems Inc.