GENESYS

9. CALL RECORDING
TABLE OF CONTENTS
9.1  Learning Objectives
9.2  Call Recording
9.3  GIR/GVP Integration – File Based Recording
9.4  3rd Party Recorder/GVP Integration – Dual Channel Rec.
9.5  Recording – NETANN vs. MSML
9.6  Recording – DN
9.7  SIP Server Initiates Call Recording
9.8  Full-time Recording
9.9  Selective Recording
9.10  Dynamic Recording – Mid-Call Control
9.11  Configuring MSML Recording
9.12  MSML Recording – SIP Server Options
9.13  Learning Check: Call Recording
9.14  Learning Check: MSML - Call Recording
9.15  Exercise: Enable Recording
9.16  MSML Recording – GVP & GIR
9.17  Exercise: DefaulT ivr Profile Recording Configuration
9.18  SIP Call Flow – MSML Recording
9.19  Exercise: Call Recording
9.20  Recording Consultation Calls
9.21  Recording During Transfers and Conferences
9.22  DTMF Clamping – Conference Recording
9.23  Call Release Tracking
9.24  Recording Alert Tone
9.25  Recording Calls with out MoH Treatment
9.26  Learning Check: Call Recording Features
9.27  Learning Check: Call Recording 2

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9.28  Exercise: Optional - Recording Features
9.29  Learning Summary
9.30  Answers

9.1  LEARNING OBJECTIVES

After completing this lesson, you will be able to:
• Identify Genesys Recording Support
οο Genesys Interaction Recording
οο 3rd Party Recorder Integration
• Describe NETANN and MSML based recording
• Configure SIP Server and GVP for MSML based recording
• Explain SIP Server recording features

9.2  CALL RECORDING

Genesys SIP Server performs call recording by passing the RTP stream
through Genesys Media Server.
Genesys Media Server acts as a proxy for the media stream and performs
one of the following:
• records the stream locally and writes the file into a storage (i.e. GIR)
• passes the stream to an external recorder (3rd party recorder
integration)

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Call recording is performed by passing a Real-Time Transport Protocol
(RTP) stream through a media server. This Media Server acts as a proxy
for the media stream, recording all media packets into a file or passing the
media stream to an external recorder.
Call recording is always enabled on a single call leg, such as with a
gateway or a SIP phone.
Call recording starts after a call becomes established. It does not result in
any changes to the call itself, to event processing, or to any other generated
TEvent.
When call recording starts, SIP Server creates two new SIP dialogs with the
media server, SIP Server sends re-INVITE requests to all call participants
with the SDP from the media server. As a result, the RTP stream between
call participants is passed via the media server.
The main role of Genesys SIP Server and GVP is to collect recording file
and metadata for a Recording Solution. Genesys offers GIR (Genesys
Interaction Recording) as the recording solution.
SIP Server and GVP also supports 3rd party recording environments such
as Verint, Zoom and Nice.

9.3  GIR/GVP INTEGRATION – FILE BASED

RECORDING

Media Server performs file based recording when integrated with GIR,
publishes the recorded file to a recording storage system (WebDAV or
Amazon S3) and then informs GIR for the new recorded interaction. GIR
collects call metadata from SIP Server via ICON.

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SpeechMiner is the end user interface for recording playback and Genesys
Quality Management. SpeechMiner also supports analytics of recorded
interactions independent from GIR.

9.4  3RD PARTY RECORDER/GVP INTEGRATION –

DUAL CHANNEL REC.
• SIP Server provides the T-Lib connection for advanced recording
controls
• Genesys Media Server relays an RTP stream per party to the 3rd
party recorder and mixes the audio so the parties hear each other

Media Server supports dual-channel call recording for third-party recorders

by extending the MSML MPCConference class. Audio only and audio-plus-
video dual-call recordings are supported, by using two SIP sessions for
each recording session, depending on the start parameter. However, Media
Server always uses an independent RTP session to stream audio for each
call in the correct order. Media Server also supports multiple recordings of
the same session.

9.5  RECORDING – NETANN VS. MSML

SIP Server supports call recording using both NETANN and MSML.
• NETANN:
οο Default and legacy method.
οο Limited capabilities / no GIR support.

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οο Supports emergency recording.
οο Wav and PCAP support.
• MSML:
οο Preferred method.
οο Advanced functionality.
οο Dynamic call recording and DN recording override.
οο Mid-call recording controls (START, STOP, PAUSE, RESUME).
οο Recording without MoH.
οο Multi-site recording operations.
οο Recording alarms.
οο Call release tracking.
οο DTMF clamping.
οο Wav and MP3 file support.
οο GIR/QM support.
SIP Server (by default) supports NETANN based recording which is the
legacy recording method. When msml is enabled for call recording SIP
Server provides advanced recording functionality.
GIR (Genesys Interaction Recording) supports msml only.

9.6  RECORDING – DN

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SIP Server uses VoIP Service DNs with service-type set to “recorder” when
NETANN is used for recording.

9.7  SIP SERVER INITIATES CALL RECORDING

SIP Server initiates call recording due to one of the following reasons:

9.8  FULL-TIME RECORDING

Enable recording through static DN configuration on either the customer
side (Trunk DN) or on the agent side (Extension DN or Agent Login).
SIP Server starts recording when the call is established on a device with
record option enabled.
οο Trunk Side Recording: Recording continues as long as the Trunk
DN remains in the call.
οο Agent Side Recording: Recording continues as long as the Agent
remains in the call.

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SIP Server starts call recording when a recording enabled device

participates into the call. When multiple recording enabled devices are
involved into a call, SIP Server creates a single recording session on
the device for which the first EventEstablished message is generated.
Recording continues for the same session as long as the device remains
in the call. If the device leaves the call and there is another device with
“record” option enabled, SIP Server terminates the initial recording session
and creates a new session.

9.9  SELECTIVE RECORDING

SIP Server starts call recording if the RequestRouteCall extension data
carries one of the following key/value pairs:
• record=source
οο Recording starts on the Trunk side.
• record=destination
οο Recording starts on the Agent side.
SIP Server overrides the DN option when:
• record=disable_destination
οο Disables the Agent side recording.
• record=disable_source
οο Disables the Trunk side recording.

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Call recording functionality can be enabled statically on a DN by setting the

record configuration option to true, or dynamically by using the record key
in the Extensions attribute of a TRouteCall request.
With this feature, call recording can be selectively disabled through a
routing strategy by overriding the record option configured on a DN. Call
recording can be disabled on either the origination DN or destination DN
when a routing strategy issues TRouteCall containing the record extension
key set to disable_source or disable_destination, respectively.
When recording is disabled by the TRouteCall request, recording can be
started on the DN by issuing a TPrivateService request after the call is
established.
DN Recording Override is supported with MSML-based call recording, for
single-site, multi-site, and Business Continuity deployments. DN Recording
Override is not supported with NETANN-based call recording.
General rules for DN Recording Override:
• If a recording configuration is overwritten for a DN, recording does
not start when a call is answered on this DN. Recording can still be
activated on this DN when the call is already established using the
TPrivateService(GSIP_RECORD_START) request.
• It is not possible to disable recording on both origination and
destination DNs using the same TRouteCall request.
• Extension key values provided in a TRouteCall request are not carried
forward to the subsequent requests.
• Call recording that is already in progress cannot be stopped.

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9.10  DYNAMIC RECORDING – MID-CALL CONTROL
Agents can control call recording through RequestPrivateService request:

Using TPrivateService requests, T-Library clients can control in real-time

an ongoing recording session. The client can pause, resume, or stop the
recording. SIP Server translates recording-related parameters from the
request to INFO messages that it sends to Genesys Media Server.
Supported mid-call actions are as follows:
• Start the recording (creates a new recording file) – SIP Server sends
MSML INVITEs to the Media Server.
• Stop the recording – SIP Server sends BYE to terminate the active
recording.
• Pause the recording – SIP Server sends INFO to pause the recording.
• Resume a paused recording – SIP Server sends INFO to resume the
recording.

9.11  CONFIGURING MSML RECORDING

Always deploy Media Server as a part of GVP for enabling Call Recording.
• RM is mandatory for enabling MSML recording.

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Enable MSML Recording on both SIP Server and GVP:
• SIP Server:
οο Enable MSML options for recording
οο Create MSML VoIP Service DN
οο Enable recording initiation (Full-time/Selective/Dynamic)
• GVP:
οο Create Recording Resource Groups
οο Configure IVR Profile options
οο Verify Media Server settings
a. MSML Section for MSML options
b. MPC Section for Media Processing options

9.12  MSML RECORDING – SIP SERVER OPTIONS

In the TServer section of the SIP Server Application object configure the
following options:
• resource-management-by-RM: true
• recording-filename: call-$ANI$-$DNIS$-$DATE$-$TIME$-$CONNID$-
$UUID$-$AGENTDN$-$AGENTID$
• msml-support: true
• msml-record-support: true
• record-consult-calls: true/false
• record-moh: false
• recording-failure-alarm-timeout: 0/<any positive integer>
• record-after-merge: true
In the Tserver section of the MSML VoIP Service DN configure the following
options – following options are required for outbound recording:
• refer-enabled: false
• make-call-rfc3725-flow: 1
• ring-tone-on-make-call: false

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9.13  LEARNING CHECK: CALL RECORDING

9.14  LEARNING CHECK: MSML - CALL RECORDING

Indicate whether each statement is true or false.

9.15  EXERCISE: ENABLE RECORDING

Objective
Configure SIP Server for MSML recording.

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Actions
• Configure SIP Server application options
• Configure geo-location on Green GW
Configure SIP Server application options
1. In GAX navigate to: Configuration > Environment > Applications >
ChicagoSIPServer > Application Options.
2. Use the Quick Filter and search for recording-filename.
3. Set the value of recording-filename option to
normal_$AGENTID$_$ANI$_$UUID$

4. Use Quick Filter and search for msml-record-support.

5. Set the value of msml-record-support option to true.

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6. Click Save.
Configure geo-location on GreenGW
7. In GAX navigate to: Configuration > Switching > DNs >
ChicagoSwitch > DNs > Green GW > Options
8. Click Add and add the following key/value pair in the TServer
section:
• geo-location = Chicago
Note: In your training environment geo-location attribute is set on both
Chicago_LRM and NewYork_LRM Resource Groups. When geo-location
is configured on the Logical Resource Groups it is mandatory to set the
geo-location attribute on the Trunk DNs for Resource Manager to provide
the recording service.

9. Click Save.
--- END OF EXERCISE ---

9.16  MSML RECORDING – GVP & GIR

GVP uses four components and functions that require additional
configuration to enable recording for GIR (GIR supports MSML only):
• Resource Manager
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• IVR Profile
• Logical Resource Group
• Media Control Platform (MCP)
For configuration option details please see Genesys Interaction Recording
Deployment Guide.
In the course exercises you will enable MSML Recording with out GIR.
The following option is mandatory on IVR Profile for the MCP to perform
MSML recording:
• [gvp.service-parameters].recordingclient.recdest: fixed,<location/
URI>

9.17  EXERCISE: DEFAULT IVR PROFILE RECORDING

CONFIGURATION
Objective
Review the MSML recording options of the default IVR Profile.
Actions
• Review the default IVR Profile configuration
Review the default IVR Profile configuration
1. In GAX navigate to: Configuration > Voice Platform > Voice Platform
Profiles > default_ivr_profile > Options.
2. Verify that the recording destination and the default file format is
configured.

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--- END OF EXERCISE ---

9.18  SIP CALL FLOW – MSML RECORDING

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Media Server performs regular call recording (conference recording) of a
two-way call by converting the call into a two-party conference. Both call
participants are re-invited to the conference call by sending the following
URL in the INVITE messages to Media Server:
sip:msml=<conf-id>@<resource-managaer>;<dn>=<DN>;record
Where the parameter specifies the file name for recording.
The following scenario describes a two-step process that is used for codec
negotiation:
1. SIP Server sends an INVITE message with the capabilities from the
original conversation with the addition of the a=inactive attribute,
in order to notify Media Server of the endpoint capabilities. Codec
negotiation for each party in the conference is done independently,
therefore, Media Server must know the endpoint capabilities in
advance, to avoid selecting different codecs for different parties,
and to minimize transcoding.
2. SIP Server sends a re-INVITE request, to which Media Server
responds in the same way that it does for a regular conference
(except that it uses only those codecs that are supported by both
participants).
Media Server configuration controls which of the following files the call-
recording service produces:
• An audio file with mixed streams (recommended)
• Separate pcap files with all of the packets captured for each stream
(when NETANN is used)

9.19  EXERCISE: CALL RECORDING

Objective
Record calls using MSML.
Actions
• Make an inbound call and start dynamic recording
Make an inbound call and start dynamic recording
1. From shortcuts open Windows Explorer and then navigate to C:\
recordings.

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2. Make sure AAgent is logged into Workspace and ready to receive

calls.
3. Using the green customer phone dial 6000.
4. The call is delivered to AAgent, accept the call on Workspace.
5. Using the Workspace call control buttons start voice recording.

6. SIP Server starts the call recording and Workspace displays the
recording indication marks on the UI.
7. Open the recordings folder and verify that the recording file has
been created.

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8. Release the call on Workspace.

--- END OF EXERCISE ---

9.20  RECORDING CONSULTATION CALLS

• SIP Server supports recording of consultation calls.
• A separate recording file is created for consultation calls.
• Enable the following options for recording the consultation calls:
οο [TServer].record-consult-calls at the SIP Server application object.
οο [Tserver].record at the agent or supervisor DN (via configuration or
the extension data)

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Regular call recording is also available for consultation calls. If enabled
for it, SIP Server starts call recording when any DN that is involved in
the consultation call is set for recording. Once recording is initiated, it
continues for as long as at least one party that is set for recording remains
in the call.
Recording ends when no more recording-enabled parties are left. For
example, if recording for both main and consultation calls is initiated by a
single party, and that party then initiates a TCompleteTransfer, recording on
both calls is terminated once the transfer is completed.

9.21  RECORDING DURING TRANSFERS AND

CONFERENCES
Recording is sticky with the device for which it has started.
Recording continues on the same file as long as the initiator device
remains in the call.
The transfer and the conference scenarios:
1. Call transfers:
• Recording continues on the same file if the recording initiator device
remains in the call after the transfer is completed.
• Recording stops if there is no “record enabled” device in the call after
the transfer is completed.
• Recording starts on a new file if there is a “record enabled” device
after the transfer is completed by the recording initiator.
2. Conference calls:
• Recording continues on the same file after the recording initiator
device starts a conference call.
• Recording stops if the recording initiator device leaves the
conference call.
• Recording starts on a new file if there is a “record enabled” device
after the recording initiator device leaves the conference call.

9.22  DTMF CLAMPING – CONFERENCE

RECORDING
When enabled SIP Server does not record the dialed DTMF digits during a
conference of an agent, a Voice Application and the customer.

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To activate DTMF clamping for recordings:
• Enable recording on agent device (not customer)
• Enable SIP Server application option: [TServer].clamp-dtmf-allowed
• Enable Trunk DN option: [TServer].clamp-dtmf-enabled

PCI compliance requires that DTMF tones are not recorded when clamping
is enabled. To satisfy this requirement, recording must be disabled on the
caller’s leg. Otherwise, DTMF digits dialed by a caller could be recorded.
Genesys recommends that you enable recording on the agent’s leg.
Configure DTMF Clamping for the conference calls where agent calls a
voice application for the customer to enter digits.
DTMF Clamping feature of SIP Server avoids the in-band DTMF digits to
be delivered to the Agent through the RTP stream and also recorded by the
media server.
For avoiding DTMF digits to be recorded during direct GVP calls refer to
MCP DTMF Record options explained in the Media Server Deployment
Guide.

9.23  CALL RELEASE TRACKING

SIP Server uses the extension data in EventRelease to provide information
about the party (customer or agent) which releases the call.
The information can be used by reporting solutions and GIR.
Enable the following option on the SIP Server application object to enable
call release tracking:
• [TServer].releasing-party-report
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In DN-based reporting, information about who released the call is

reported in the AttributeExtension using extension key ReleasingParty
in EventReleased and EventAbandoned events, when those events are
distributed.
SIP Server includes one of the following values in the ReleasingParty key:
• 1 Local: The call is released because the ThisDN value in the
EventReleased requested the release.
• 2 Remote: The call is released because the other party (which is
remote to ThisDN) in the EventReleased or EventAbandoned events
requested the release.
• 3 Unknown: The call is released, but SIP Server cannot determine the
release initiator.

9.24  RECORDING ALERT TONE

Configure the following option on the default IVR Profile for enabling the
Record Alert Tone:
• [gvp.service-parameters].recordingclient.audiosrc = fixed,<url>
• [gvp.service-parameters].recordingclient.tonesilenceduration =
fixed,<integer>
SIP Server does not record the recording alert tones.
To meet the regulatory requirements, some deployments require the
system to periodically generate an audio tone to notify the participants in a
call that the call is currently being recorded. The following options describe
how to work with audio tones:

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audiosrc:
The URI of the audio tone. If the URI is set to an empty string, or not
defined, or resolves to a bad URI, then no audio tone is applied to the call.
No other notifications are generated by Media Server (for example, MSML
events) when no audio tone is applied.Only .wav files are supported. QTMF
tones and files stored in sub-directories with multiple codecs that are
supported by Media Server are not supported. For example, “music/beep”
cannot be specified for this option, even though it is valid for other Media
Server treatments.
tonesilenceduration:
The length of time, in milliseconds, between playing the audio tone.
This is a mandatory parameter if the Recording Alert Tone parameter is
defined, otherwise no audio tone is applied. The minimum accepted value
is 1500 (if a smaller value is specified, 1500 will be used). In addition, if
the Frequency of Recording Alert Tone parameter is not present, MCP
applies the default value of 30000 instead of not applying tone.

9.25  RECORDING CALLS WITH OUT MOH

TREATMENT
• SIP Server provides the ability to record calls without MoH treatment.
• Disable the [TServer].record-moh option on SIP Server to stop
recording the MoH treatments.

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9.26  LEARNING CHECK: CALL RECORDING
FEATURES
Indicate whether each statement is true or false.

9.27  LEARNING CHECK: CALL RECORDING 2

Indicate whether each statement is true or false.

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9.28  EXERCISE: OPTIONAL - RECORDING
FEATURES
Objective
Practice SIP Server recording features.
Actions
• Disable record-moh and enable call release tracking
• Enable Recording Alert Tone
• Record a call and analyze SIP Server logs
Disable record-moh and enable call release tracking
1. In GAX navigate to: Configuration > Environment > Applications >
ChicagoSIPServer > Application Options.
2. Use the Quick Filter and search for record-moh.
3. Set the value of the record-moh to false.

4. Use Quick Filter and search for releasing-party-report.

5. Set the value of the releasing-party-report to true.

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6. Click Save.
Enable recording alert tone
7. In GAX navigate to: Configuration > Voice Platform > Voice Platform
Profiles > default_ivr_profile > options.
8. Click Add.
9. Add the following key/value pairs in the gvp.service-parameters
section:
• recordingclient.audiosrc = fixed,file:///recordings/beep.wav
• recordingclient.tonesilenceduration = 2000

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10. Click Save.

Record a call and analyze SIP Server logs
11. Make sure AAgent is logged into Workspace and ready to receive
calls.
12. Using the green customer phone call 6000.
13. The call is delivered to AAgent, accept the call.
14. Start voice recording on Workspace.
15. Verify that the recording alert tone - beep is played back to the call
participants.
16. Using Workspace put the call on hold.
17. In SIP Log Visualizer reload the SIP log file.
18. Go to the bottom of the ladder diagram.
19. Identify the last RequestHoldCall.
20. SIP Server sends INFO to the Media Server to pause MOH
recording.
21. Identify the INFO and open the message.
• The second INFO after the RequestHoldCall.
22. Verify the pause command in the MSML section in INFO.

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23. Release the call on Workspace.

24. In SIP Log Visualizer reload the log file.
25. Identify the last RequestReleaseCall message.
26. Identify the first EventReleased message that SIP Server generated
after receiving RequestReleaceCall.
27. Open the EventReleased message and verify the releasing party
from the extension data.

--- END OF EXERCISE ---

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9.29  LEARNING SUMMARY
Now that you have completed this lesson, you should be able to:
• Identify Genesys Recording Support
οο Genesys Interaction Recording
οο 3rd Party Recorder Integration
• Describe NETANN and MSML based recording
• Configure SIP Server and GVP for MSML based recording
• Explain SIP Server recording features

9.30  ANSWERS
Call Recording:
A, C, D, E.
MSML Call Recording:
1. True
2. True
3. True
Call Recording Features:
1. False - Call release tracking is a SIP Server feature.
2. False - Configure SIP Server to avoid MoH recording.
3. False - DTMF clamping is required only when confidential
information is provided by customer.

Call Recording 2:
1. False - Recording alert tone is played with specific intervals.
2. True
3. False - Consultation calls are recorded into a separate file.

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