Genesys Application Note

AudioCodes Mediant SBC

With Genesys SIP Server
Document version 1.1
The information contained herein is proprietary and confidential and cannot be disclosed
or duplicated without the prior written consent of Genesys, Inc.

Copyright © 2014-2021 Genesys, Inc. All rights reserved.

About Genesys
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by Genesys Cloud™, an all-in-one solution and the world's leading public cloud contact center platform,
designed for rapid innovation, scalability and flexibility. Go to www.genesys.com for more information.

Notice
Although reasonable effort is made to ensure that the information in this document is complete and
accurate at the time of release, Genesys Telecommunications Laboratories, Inc. cannot assume
responsibility for any existing errors. Changes and/or corrections to the information contained in this
document may be incorporated in future versions.

Your Responsibility for Your System’s Security

You are responsible for the security of your system. Product administration to prevent unauthorized
use is your responsibility. Your system administrator should read all documents provided with this
product to fully understand the features available that reduce your risk of incurring charges for
unlicensed use of Genesys products.

Trademarks
Genesys and the Genesys logo are registered trademarks of Genesys Telecommunications Laboratories,
Inc. All other company names and logos may be trademarks or registered trademarks of their
respective holders.

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Table of Contents
1 Summary .................................................................................................................................... 4
2 Feature Support .......................................................................................................................... 5
3 Software and Hardware Versions Validated .................................................................................. 9
4 Functional Test Case Scenarios .................................................................................................. 10
5 Features Configuration in Genesys Configuration Environment .................................................... 11
6 Gateway/SBC Configuration....................................................................................................... 14

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1 Summary

AudioCodes Mediant Session Border Controller (SBC) is recommended for integration with the Genesys
SIP solution.

As noted in section 2 and 4 below, all test calls/cases were successfully executed.

This application note is applicable to the following AudioCodes products:

 Mediant 500 SBC
 Mediant 800B SBC
 Mediant 1000B SBC
 Mediant 2600 SBC
 Mediant 3000 SBC
 Mediant 4000 SBC
 Mediant 9000 SBC
 Mediant Software and Virtual SBC

The supporting versions of Genesys components include SIP Server 8.1.1, SIP Feature Server 8.1.2,
Media Server (8.1.x and 8.5.x), and SIP Proxy v8.1.1.

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2 Feature Support
2.1 General Features

SIP Trunk or Gateway - Description Supported Test Cases

Feature Compatibility
Inbound Calls - Standard Direct calls to a phone/user with a DID # Yes 1,2,3,4,12
Contact Center calls; may be queued or played
Inbound Calls - Contact Center
some announcements before being routed to an Yes 5,6,7,13,22,23
/ Routed
agent
Manually Dialed, or Forwarded to external
Outbound Calls - Standard Yes 9,10,11
destination
Outbound Calls - Automated Automated dialing by Genesys OCS or similar
Dialer Campaign, CPD by application Call Progress Detection (CPD) by Yes 25
Genesys Genesys Media Server*
Remote Agent, not Registered Typically using a PSTN phone behind the
Yes 24
to SIP Server gateway or SIP Trunk
Simple IVR controlled by a routing strategy, and
Call Qualification & Parking Yes 5,6,7,22,23
queuing of calls with announcements or music
GVP – Advanced IVR (VXML,
TTS, ASR, etc), Conferencing, & Same SIP signaling as qualification & parking Yes 6,7
more
No dedicated test
Call Recording No meaningful impact to SIP signaling
cases

* CPD may also be performed by the gateway if it returns results in a format compatible with Genesys
SIP. Please note such capabilities if they are available.

Note: Support for Answering Machine Detection (AMD) and Call Progress Tone Detection (CPD) on the
Mediant 4000, Mediant 9000, and the Mediant Software and Virtual SBC is planned for the AudioCodes
7.0 software release. It is supported on other AudioCodes SBC devices.

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2.2 Technical Features

Technical Compatibility –
Description Supported Test Cases
Architecture & SIP Protocol
All test cases
“Single Site” One instance of Genesys SIP Server Yes
apply
Two or more instances of Genesys SIP Server, No “dedicated”
“Multisite”
behind a single Trunk and/or SBC test cases

Transfer method reflects the signaling sent to the

Transfer with re-INVITE Yes 14,15
SIP Trunk or gateway
Transfer with 3xx Redirect prior to call connection Yes 8
Transfer method reflects the signaling sent to the
Transfer with REFER Yes 16,17,19,20,21
SIP Trunk or gateway
Conference controlled on Genesys SIP Server &
Ad Hoc Conference Yes 18
Media Server
SIP Authentication Yes 27, 28
See the Genesys 8.1 SIP Server Deployment Guide No dedicated
SIP Over TLS Yes
for details. test cases
No dedicated
SRTP Yes
test cases
Service Monitoring Monitoring with OPTIONS messages Yes 26

SIP Server High Availability - No dedicated

Effectively transparent to external devices Yes
with Virtual IP Address test cases
SIP Trunk/SBC/Gateway High
No dedicated
Availability - with Virtual IP Effectively transparent to external devices Yes
test cases
Address
Support for a highly available SBC or SIP Trunk
[not tested –
SIP Trunk/SBC/Gateway High with either multiple active nodes or Not covered
requires
Availability – List of IP primary/backup; SIP Server is configured with the by standard
supplemental
Addresses IP address of each node (typically using the test plan
testing]
backup contact setting on SIP Server)
[not tested –
SIP Server High Availability - Not covered
Architectures with SIP Proxy used to manage high requires
DNS-based Redundancy with by standard
availability supplemental
SIP Proxy test plan
testing]
[not tested –
SIP Trunk/SBC/Gateway High Support for an SBC or SIP Trunk with DNS-based Not covered
requires
Availability - DNS-based redundancy (the contact of the DN on SIP Server by standard
supplemental
Redundancy would be hostname/FQDN) test plan
testing]
All test cases
The test plan does not include dedicated tests for utilize the
Audio Codec Support each codec; codecs are supported by Media Yes “negotiated
Server/GVP, and by the SIP endpoints preferred”
codec
[not tested –
The test plan does not include dedicated tests for
requires No dedicated
Video Support video; video is supported by Media Server/GVP,
supplemental test cases
and by the SIP endpoints
testing]

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2.3 SBC-specific Features

SBC Feature Compatibility for

Agent REGISTERed to SIP Server Description Supported Test Cases
through SBC
Inbound & Outbound Calls Yes 29,30
SIP Agent 3PCC Control Yes 29
Remote Agent - Transfer with
Yes 30
REFER (SIP Phone via SBC)
Transfer with REFER Yes 30
Transfer with reINVITE Yes 31

2.4 Details Regarding Features

2.4.1 Multisite
Note:
This application
SIP Trunk or SIP Trunk or note uses the term
Gateway Gateway “multisite” to cover
architectures with
transfers with ISCC,
which conform to
either option on the
SBC left: a SIP
Trunk/Gateway
through a single
SBC, or a SIP
Trunk/Gateway
connected directly.
SIP Server 1 SIP Server 2 SIP Server 1 SIP Server 2
or or Either REFER or
T-Server + T-Server + reINVITE may be
PBX PBX tested and
supported.

Architectures with 2
or more SBCs are
beyond the scope of
this app note.

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2.4.2 High Availability

This Application Note and the Test Plan provide coverage and support for High Availability
accomplished with a “Virtual IP Address.” This is also referred to as “IP Address Takeover” or a
“Floating IP Address.”

The general approach is that the “active” instance of a component utilizes this special IP address. It is
typically transparent at the SIP signaling layer which instance is active. A Genesys SIP Server, an SBC,
or the components may employ this high availability on the interface for a standard “SIP Trunk.”

Other methods of high availability do exist. These methods require more advanced logic on the part of
each SIP component to monitor multiple instances of another component, and select the appropriate
instance.

For example, SIP Server supports configuring a primary and back IP address for a component (using
the contact and contacts-backup options). This type of method is referred to as a “list of IP Addresses”
in this application note. In another example, a SIP Server does support using an FQDN to reach
another component, and can utilize multiple DNS records to help choose the best component instance.
This method is referred to as “DNS-based HA.”

Both the “List of IP Addresses” and DNS-based high availability methods are beyond the scope of this
Application Note (and this limitation applies in both directions, from SIP Server towards an external
component, and vice versa from an external component towards SIP Server).

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3 Software and Hardware Versions Validated

The following Genesys components and AudioCodes SBC were validated for reference configuration
examples.

3.1 Genesys Components

Genesys Components

Component Version Notes

Genesys SIP Server performs call

switching and control. SIP Server
SIP Server 8.1.1
communicates via SIP with SIP
Endpoints.

Used to handle media interactions

such as call treatments (ring back,
Genesys Media Server 8.1.700
busy tones and music on hold); also
used as MCU.

Genesys SIP Feature Server 8.1.2 Used as a SIP Voicemail Server

Optionally can be used for DNS-

SIP Proxy 8.1.1
based HA deployment

3.2 Gateway/SBC

3rd Party Hardware Components

Model Version Notes

AudioCodes Mediant SBC 6.8

For a full listing of 3rd party hardware/software supported by Genesys, see the Genesys Supported
Media Interface Guide (SMI) and the SIP Integration Reference.

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4 Functional Test Case Scenarios

Functional Test Cases

# Scenario Description Supported
1 Inbound Call to Agent released by caller Yes
2 Inbound Call to Agent released by agent Yes
3 Inbound Calls rejected Yes
4 Inbound Call abandoned Yes
5 Inbound Call to Route Point with Treatment Yes
6 Interruptible Treatment Yes
7 IVR (Collect Digit) Treatment Yes
8 Inbound Call routed by using 302 out of SIP Server signaling path Yes
9 1PCC Outbound Call from SIP Endpoint to external destination Yes
10 3PCC Outbound Call to external destination Yes
11 1PCC Outbound Call Abandoned Yes
12 Caller is put on hold and retrieved by using RFC 2543 method Yes
13 T-Lib-Initiated Hold/Retrieve Call with MOH using RFC 3264 method Yes
14 3PCC 2 Step Transfer to internal destination by using re-INVITE method Yes
15 3PCC Alternate from consult call to main call Yes
16 1PCC Unattended (Blind) transfer using REFER Yes
17 1PCC Attended Transfer to external destination Yes
18 3PCC Two Step Conference to external party Yes
19 3PCC (same as 1PCC) Single-Step Transfer to another agent Yes
20 3PCC Single Step Transfer to external destination using REFER Yes
21 3PCC Single Step Transfer to internal busy destination using REFER Yes
22 Early Media for Inbound Call to Route Point with Treatment Yes
23 Early Media for Inbound Call with Early Media for Routed to Agent Yes
24 Inbound call routed outbound (Remote Agent) using INVITE without SDP Yes
25 Call Progress Detection Yes
26 Out of Service detection; checking MGW live status Yes
27 SIP Authentication for outbound calls Yes
28 SIP Authentication for incoming calls Yes
SBC-Specific Test Cases
T-Lib-Initiated Answer/Hold/Retrieve Call for Remote SIP endpoint which supports the Yes
29
BroadSoft SIP Extension Event Package
30 3PCC Outbound Call from Remote SIP endpoint to external destination Yes
31 3PCC 2 Step Transfer from Remote SIP endpoint to internal destination Yes
32 1PCC Attended Transfer from Remote SIP endpoint to external destination Yes

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5 Features Configuration in Genesys Configuration Environment

Genesys SIP Configuration

Features Supported By Gateway/SBC

Feature Key Actions and Procedures

The following is the default/standard SIP Server configuration used during testing of the technical
features:

SIP Server Application Options TServer section

sip-hold-rfc3264=true
router-timeout=30
default-dn=
blind-transfer-enabled=true
resource-management-by-rm=true
msml-support=true
sip-enable-moh=true

The following is the default/standard for DN configuration used during testing of the technical features:
Name Number Name in CME Options TServer section Comment
CME
MGW- MGW- MGW- refer-enabled=true TSE
TRUNK TRUNK TRUNK contact=<TSE_CONTACT>
oos-check=10
oos-force=5
oosp-transfer-enabled=true
sip-replaces-mode=2
Ext-DN1 21001 N/A N/A
Ext-DN2 21002
SIP-DN1 7101 7101 refer-enabled=false
SIP-DN2 7102 7102 ring-tone-on-make-call=false
make-call-rfc3725-flow=1
contact=*
SIP-RDN 7200 7200 refer-enabled=true SIP endpoint which
ring-tone-on-make-call=false supports the BroadSoft SIP
make-call-rfc3725-flow=1 Extension Event Package.
contact=*
sip-cti-control=talk,hold
SVC_MSML SVC_MSM SVC_MS prefix=msml= MS
L ML contact=<MS_CONTACT>
service-type=msml
subscription-id= Environment

1, 2, 3, 4,
5, 6, 7, 9,
Use the default/standard configuration indicated above
11, 13, 18,
30, 31

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 In addition to the default/standard configuration indicated above, set the following:
8
oosp-transfer-enabled=true

In addition to the default/standard configuration indicated above, set the following:

10
refer-enabled=true

In addition to the default/standard configuration indicated above, set the following:

12
sip-hold-rfc3264=false

In addition to the default/standard configuration indicated above, set the following:

14
refer-enabled=false

In addition to the default/standard configuration indicated above, set the following:

15
refer-enabled=true

In addition to the default/standard configuration indicated above, set the following:

16
refer-enabled=true

In addition to the default/standard configuration indicated above, set the following:

17
oosp-transfer-enabled=true

In addition to the default/standard configuration indicated above, set the following:

19
refer-enabled=true

In addition to the default/standard configuration indicated above, set the following:

20 refer-enabled=true
oosp-transfer-enabled=true

In addition to the default/standard configuration indicated above, set the following:

21 refer-enabled=true
sip-busy-type=2

In addition to the default/standard configuration indicated above, set the following:

22
sip-early-dialog-mode=1

In addition to the default/standard configuration indicated above, set the following:

23
sip-early-dialog-mode=1

In addition to the default/standard configuration indicated above, set the following:

24
oosp-transfer-enabled=false

In addition to the default/standard configuration indicated above, set the following:

25 cpd-capability = mediaserver
refer-enabled=false

In addition to the default/standard configuration indicated above, set the following:

26 oos-check=10
oos-force=5

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 In addition to the default/standard configuration indicated above, on the Annex tab, configure the
27
AuthClient section with options username=<username> password=<password>

In addition to the default/standard configuration indicated above, set the following:

28
authenticate-requests=invite
password=1234

In addition to the default/standard configuration indicated above, set the following:

sip-cti-control=talk,hold
29
authenticate-requests=REGISTER
password=1234

In addition to the default/standard configuration indicated above, set the following:

32
refer-enabled=false

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6 Gateway/SBC Configuration

This section describes how to configure features represented in the Feature Support (see
section 2).

Gateway/SBC Configuration

Features Supported By Gateway/SBC

Feature Key Actions and Procedures

The following sections taken in order describe the steps necessary to configure basic SBC
functionality for an AudioCodes SBC device. These sections are provided only as a reference. The
user should refer back to respective area within the User’s Manual for the particular AudioCodes
device for complete details and explanations of all the options.

1. Network (LAN & WAN)

a. Define the network interface for the Genesys (Trusted or LAN) Network. Reset is required.
Configuration > VoIP > Network > IP Interfaces Table

1, 2, 3, 4, 5, 6,
7, 8, 9, 11, 12,
13, 14, 18, 22,
23, 24, 25, 26,
27, 28, 29, 31

Define the network interface for the ITSP (Untrusted or WAN) Network. Reset is required.
Configuration > VoIP > Network > IP Interfaces Table

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 2. SBC Application Enabling
Enable the SBC Application (for a device which isn’t a ‘pure’ eSBCs). Reset is required.
Configuration > Applications Enabling

3. Media Realm
Define Media Realms for the Media interfaces (both Genesys & ITSP Provider).
Configuration > VoIP > Network > Media Realm Table

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 4. SRDs
Define SRDs for the networks (both Genesys & ITSP Provider).
Configuration > VoIP > Network > SRD Table

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 5. SIP Interfaces
Define SIP Interfaces.
Configuration > VoIP > Network > SIP Interface Table

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 6. IP Groups
Define the supporting IP Groups.
Configuration > VoIP > Network > IP Group Table

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 7. Proxy Sets
Define Proxy Sets.
Add the address(es) or FQDN of the Genesys SIP Server(s) to the Proxy Set assigned to the IP
Group in the previous step.
Note: Proxy Set ID 0 (zero) is reserved as a ‘default proxy’ and should not be used with IP groups.
Note: While a single proxy is shown below, a Proxy Set should be created for each defined IP Group.
Configuration > VoIP > Network > Proxy Sets Table

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 8. IP-to-IP Routing
Define any required IP-to-IP Routing rules.
For this example (associated rule highlighted for each item):
0. OPTIONS are terminated at the SBC.
1. All calls from ITSP will route to Genesys SIP Server.
2. All calls from Genesys environment will route to the ITSP.
Configuration > VoIP > SBC > Routing SBC > IP-to-IP Routing Table

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 9. Digit/SIP-URI Manipulation
Define any required IP-to-IP Inbound/Outbound Manipulations.
In this example, the leading “+” is stripped from Destination numbers from Genesys toward the ITSP
as needed.
Configuration > VoIP > SBC > Routing SBC > IP-to-IP Inbound Table

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 10. Message Manipulation
Define any required Message Manipulations.
An explanation for this example:
1. The ITSP sends new INVITES when a SIP 603 Declined response is returned on an initial
INVITE.
2. The ITSP does not send the new SIP INVITES for SIP 600 Busy Everywhere response.
In this case, the header.request-uri.methodtype changes the ‘603’ to ‘600’ response code preventing
new SIP INVITEs for the same call.
Note: The inbound/outbound manipulation set identifier should be configured against the appropriate
IP Group in the IP Group table. In this case, “Outbound Message Manipulation Set” for IP Group 2
(for the ITSP/Provider) should be configured to ‘1’.
Note: Depending upon the ITSP/Provider, the need may exist to implement several message
manipulations to include a Diversion header or to modify the Contact header (the later in cases where
external callers are being referred out to the network) Additionally, there may exist, depending upon
the customer, manipulation rules to do topology hiding.
Configuration > VoIP > SIP Definitions > Msg Policy & Manipulation > Message Manipulations

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 Taking into account the information provided above when configuring basic SBC functionality for an
AudioCodes SBC device, the following sections would be followed when the SBC interfaces an ITSP
which does not support use of SIP REFER. These sections are provided only as a reference. The
user should refer back to the respective area within the User’s Manual for the particular AudioCodes
device in question for complete details and explanations of all options.
Note that by default the SBC device’s handling of SIP 3xx redirect responses is to send the Contact
header unchanged. However, some SIP entities may support different versions of the SIP 3xx
standard while others may not even support SIP 3xx.
10,15,16,17,19, For ITSPs, if SIP REFER isn’t handled, they usually do not support proper SIP 3xx behavior and
20,21,30 require equal treatment. That means being handled locally by the AudioCodes SBC with a trigger to
direct a SIP INVITE be sent to the ITSP and the call anchored on the SBC (but SIP Server is
released from the call). See below:

1. IP Profile
Configure the IP Profile associated to the ITSP to have SIP REFER and 3xx responses handled
locally by the SBC.
Configuration > VoIP > Coders and Profiles > IP Profile Settings > Select the IP Profile
associated to the SIP Trunk Provider > select the SBC tab

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 2. IP-to-IP Routing

Establish routing rule(s) for routing the SIP re-INVITE to the ITSP if the triggering message was a SIP
REFER.

Configuration > VoIP > SBC > Routing SBC > IP-to-IP Routing Table > Add route to get to SIP
Trunk provider

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 11. Answering Machine Detection (AMD)/Call Progress Tone Detection (CPD)
Note: Support for Answering Machine Detection (AMD) and Call Progress Tone Detection (CPD) on
the Mediant 4000, Mediant 9000, and the Mediant Software and Virtual SBC is planned for the
AudioCodes 7.0 software release.
AudioCodes Media Gateways support answering machine detection (AMD) as well as call progress
tone detection capabilities that can detect whether a human voice or an answering machine is
answering the call. This capability is useful for automatic dialing applications.
To enable and configure AMD:
1. Using the Media Gateway web interface, open the IPMedia Settings page (Configuration tab >
VoIP > Media > IPMedia Settings):

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 2. Enable AMD by setting the 'IPMedia Detectors' parameter to Enable.
3. Configure the other AMD parameters as required. See below for a description.
4. Click Submit then save (burn) the setting to flash memory with a device reset.
5. To enable voice detection once the AMD detects the answering machine, set the ini file
parameter, EnableVoiceDetection to 1.
The Media Gateway supports up to four AMD parameter suites, where each parameter suite defines
the AMD sensitivity levels of detection. The sensitivity levels can range from 0 to 15, depending on
the parameter suite. The level is selected using the 'Answer Machine Detector Sensitivity Level'
parameter (AMDSensitivityLevel). The parameter suite(s) can be loaded to the device in the Web
interface as an auxiliary file or remotely through the ini file using the AMDSensitivityFileName and
AMDSensitivityFileUrl parameters.
Additionally AMD can also be configured per call based on the called number or Trunk Group. This is
done by configuring AMD for a specific IP Profile and then assigning the IP Profile to a Trunk Group
in the Inbound IP Routing table.
The Media Gateway also supports the detection of beeps at the end of an answering machine
message. This allows users of third-party,application servers to leave voice messages after an
answering machine plays a “beep” sound.
The Media Gateway supports the following methods for detecting and reporting beeps:
 Using the AMD detector: This “beep” detector is integrated in the existing AMD feature. The
beep detection timeout and beep detection sensitivity are configurable using the
AMDBeepDetectionTimeout and AMDBeepDetectionSensitivity parameters, respectively. To
enable the AMD beep detection, the X-Detect header in the received SIP INVITE message
must include “Request=AMD”, and the AMDBeepDetectionMode parameter must be set to 1
or 2. If set to 1, the beep is detected only after Answering Machine detection. If set to 2, the
beep is detected even if the Answering Machine was not detected.
 Using the Call Progress Tone detector: To enable this detection mode, the X-Detect
header in the received SIP INVITE message must include “Request=CPT”, and one or
several beep tones (Tone Type #46) must be configured in the regular CPT file.
The Media Gateway reports beep detection by sending a SIP INFO message containing a body with
one of the following values:
 Type=AMD and SubType=Beep
 Type=CPT and SubType=Beep

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 Upon AMD activation, the device can send a SIP INFO message to an application server notifying it
of one of the following:
 Human voice has been detected
 Answering machine has been detected
 Silence (i.e., no voice detected) has been detected
The detected AMD type (e.g., voice) and success of detecting it correctly are also sent in CDR and
Syslog messages.

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