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Document Overview

This document provides a configuration guide for Ribbon Session Border Controller (SBC) Q21 series when connecting with the Genesys Voice Platform.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Ribbon SBC Q21 and Genesys Voice Platform.

Audience

This technical document is intended for telecommunication engineers with the purpose of configuring the Ribbon SBC Q21 series  with the Genesys Voice PlatformThis configuration requires the navigation of a third-party server and the Ribbon SBC Q21 Command Line Interface (CLI). Understanding the basic concepts for IP/Routing, SIP, RTP, and TLS are also required for completing the configuration and any necessary troubleshooting.

Requirements

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

 

Equipment

Software Version

Ribbon Communications

SBC Q21

9.3.3.0

 

Third-party Equipment



Genesys SIP Server8.1.100.98

Genesys GVP

8.1.504.93

Polycom SoundPoint IP 501 SIP

2.1.3

Reference Configuration

The following reference configuration shows connectivity between third-party and Ribbon Q21.

Figure : Topology


Support

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

 

Third-Party Product Features

Inbound Calls 

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

Outbound Calls

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

Music on hold

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

Call Transfers

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

Other scenarios

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

Remote Agent behind SBC scenarios

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

Genesys Voice Platform configuration

The following new configurations are included in this section:

  1. SIP Server Configuration

  2. SIP Trunk Configuration
  3. SIP Extension
  4. Route Point
  5. IVR Profile

  6. SIP DN Configuration
  7. EpiPhone Configuration

1. SIP Server Configuration

 

The following configuration shows how objects are configured by default:

 

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

2. SIP Trunk Configuration

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

Figure : SIP Trunk Configuration

 

3. SIP Extension  

Under SIPSwitch, define the DNs Extension types with the following options in the TServer section for the various SIP endpoints. These will register to the SIP Server.

 

Figure : SIP Extension

 

 

4. Route Point

In SIPSwitch, create a DNs Routing Point type. This type should match the Request URI user part.

Figure : Route Point

 

5. IVR Profile

The following displays the content from the configuration file esttt.conf:

Figure : IVR Profile

6. SIP DN Configuration

 

Table : SIP DN Configuration

Name

Number

Name in CME

CME Options TServer section

Comment

MGW-TRUNK

MGW-TRUNK

MGW-TRUNK

refer-enabled=true

contact=<TSE_CONTACT>

oos-check=10

oos-force=5

oosp-transfer-enabled=true

sip-replaces-mode=2

TSE

Ext-DN1

Ext-DN2

2144326886

2144326887

N/A

N/A

 

SIP-DN1

SIP-DN2

SIP-DN3

2086041020

2086041021

2086041022

1020

1021

1022

refer-enabled=false

ring-tone-on-make-call=false

make-call-rfc3725-flow=1

contact=*

 

SIP-RDN

2086041022

1022

refer-enabled=true

ring-tone-on-make-call=false

make-call-rfc3725-flow=1

contact=*

sip-cti-control=talk,hold

SIP endpoint which supports the BroadSoft SIP Extension Event Package.

RP

20860410111011

 

 

RP1

2086041012

1012

 

 

RP2

2086041013

1013

 

 

SVC_MSML

SVC_MSML

SVC_MSML

prefix=msml=

contact=<MS_CONTACT>

service-type=msml

subscription-id= Environment

MS

 

7. EpiPhone Configuration

Content of configuration file esttt.conf.

EpiPhone Configuration
[TcCM]
site1 = UTE_HOME
connect-on-startup  = true
open-log-on-startup = false
log-to-file = epi-phone.log

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

[UTE_HOME]
server = (host=<SIP_SERVER_HOST_IP>,port=<SIP_SERVR_TLIB_PORT>)

sip-register = false
dn8 = 2086041020,name="DN1",mkcall="2086041021"
dn9 = 2086041021,name="DN2"
dn15 = 2086041022,name="DN3"
dn25 = 2076041025,name="RDN"
dn10 = 1011,pool="shared"
dn11 = 1012,pool="shared",script="annc=(PROMPT=(\"1\"=(INTERRUPTABLE=1,ID=1)))"
dn12 = 1013,pool="shared",script="collect=(MAX_DIGITS=4,RESET_DIGITS=11,BACKSPACE_DIGITS=22,TOTAL_TIMEOUT=1000) play=music/on_hold"

 

SBC Q21 Configuration

 

SBC Q21 Configuration
#Basic Q21 Configuration
cli vnet add public
cli vnet edit public ifname eth2 gateway 10.35.141.65

cli vnet add private
cli vnet edit private ifname eth3 gateway 10.35.141.97

cli realm add public1
cli realm edit public1 rsa 10.35.141.94 mask 255.255.255.224 vnet public
cli realm edit public1 medpool 1 imr on emr on
cli realm edit public1 sipauth all

cli realm add private1
cli realm edit private1 rsa 10.35.141.124 mask 255.255.255.224 vnet private
cli realm edit private1 medpool 2 imr on emr on
cli realm edit private1 sipauth all

cli iedge add Genesys 0
cli iedge edit Genesys 0 type sipproxy realm private1 static 10.35.176.111 dtg togen newsrcdtg gentrunk
cli iedge edit Genesys 0 invitenosdp yes
cli iedge edit Genesys 0 uri 10.35.176.111

cli iedge add gen_trunk 0
cli iedge edit gen_trunk 0 realm public1 type sipgw static 10.35.8.146 newsrcdtg togen dtg gentrunk
cli iedge edit gen_trunk 0 removeVIDEOfromSDP enable
cli iedge edit gen_trunk 0 invitenosdp yes

cli realm edit public1 proxy_regid Genesys proxy_uport 0

#Message manipulation rule which changes m-line with video for SIP 200OK
cli fmm trigger add blankvideo-mline-t sdp-body
cli fmm trigger edit blankvideo-mline-t code is(200)
cli fmm trigger edit blankvideo-mline-t sdp.section.type is("media")
cli fmm trigger edit blankvideo-mline-t sdp.line.type is("m")

cli fmm action add modmline modify
cli fmm action edit modmline blankvideo-mline-t.sdp.line.value substitute(blankvideo-mline-t.sdp.line.value,"(video 0 RTP/AVP)","$1 34")

cli fmm rule add modvideo
cli fmm rule edit modvideo condition while(blankvideo-mline-t) actions modmline

cli fmm profile add modvidp
cli fmm profile edit modvidp rules modvideo


 

Test Results

 

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

Conclusion

These Application Notes describe the configuration steps required for Ribbon SBC Q21 to successfully interoperate with Genesys Voice Platform. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.