PROOF OF CONCEPT

Conference Bride POC With XOP

Thilan Jayathilake
[email protected]
Version Author Date

1 Ebin Baby 03/02/2023

Company Role POC Status Signature Date

Completed
Ebin Baby Speakerbus Cloud DevOps Successfully 24/02/2023
Solutions Engineer –
Cadence
Community
Completed
Thilan Jayathilake Speakerbus Solutions Team Successfully 24/02/2023
Solutions Lead - Cadence
Community

Ahmed Mousa Speakerbus Snr. Network Completed 27/02/2023

Solutions Engineer – Successfully
Cadence
Community
Introduction
This is a generic test plan and outlines the feature set that should be tested to confirm interoperability with
third party vendors. The following bulleted list details the areas that will be addressed during the testing. In
all testing the functionality and associated behaviour along with the quality of the audio.

Firmware and Software Revisions

The following tables give details of the software and hardware used during the testing with the relevant
revisions of these pieces of software and hardware.

Firmware Product Firmware / Hardware Version

ARIA v2.200.6.0

iCMS v3.951.1.0

iCS 2.800.9.0

Audicodes SBC v.7.40A.005.306

Cisco Gateway Cisco- 3925 Gateway/IOS-15.2.3.T2

Equipment Requirements
The following bulleted list details the equipment that should be available when conducting the 3rd party
interoperability testing.
• iManager Communication Server (iCS).
• At least 2 x SB Turrets (ARIA) registered to iCS
• SIP Trunk connectivity from ICS
• SIP Private wire with Cadence Community through Cisco gateway.
Topology

10.143.11.1

10.208.14.44

10.208.14.98

10.143.37.11

Scope of the Test

ARD • Make an ARD call from A1 to B1 and B1 Release.

• Make an ARD call from A1 to B1 and A1 Release.
• Make an ARD call from A1 and A2 on the same site barge into the call, A1 barge
out and B1 Release.
• Make an ARD call from A1 and A2 on the same site barge into the call, A2 barge
out and B1 Release.
• Make ARD call from A1 and A2 barge in B1 Release.
• Make ARD call from A2 and A1 on the same site barge in B1 Release.
• A2 making an ARD A1 barge in B1 Release.
• Make ARD call from A2 and B1 Release.
• B1 making an ARD A1 answering and A1 Release.
• B1 making an ARD A1 answering A2 barge in and B1 Release.
• B1 making an ARD A1 Answering and A2 barge in.
• B1 making an ARD A2 Answering A1 barge in B1 Release.
• Call hold tests
o A1 making an ARD with B1 and A1 put B1 on hold.
 o A1 making an ARD with B1 and A1 place the call on hold A2 barge
in, A2 release the call.
HOOT • Make a Hoot call from one site to another (from site 1 to 2)
• Make a Hoot call from one site to another (from site 2 to 1)
• Two calls from site 1 to 2
• Two calls from site 2 to 1
• Second user barges into hoot on site 1
• Second user barges into hoot on site 2
• Second user can barge into two active calls on site 1
• Second user can barge into two active calls on site 2

MRD Tested, without signalling. Signalling embedded in the RTP in-band.

Sip Private Out of scope

wire High
Availability

Automatic Ring Down (ARD) Voice Services

The ARD from cadence community is implemented as an On Demand private wire this means it is only established
when a trader at one side or the other seizes the line. The ARD always does far party clear so when either side hangs
up the Speakerbus(SB) network will send BYE and clear the call. This means that if the line will be used with speakers
they will drop unless configured to not drop locally in the turret system.

HOOT Voice Services

The HOOT from SB is implemented as a Permanent private wire, and it will be established at system start up and will
normally be up all the time. The SB network expects the turret system to send the INVITE to start the call as the
network is in effect in answer only mode for permanent private wires.

Configuration steps

Step 1: Configuring a SIP Trunk to Audio Codes SBC

Navigate to PBX > ICS>SIP Trunk, as shown in the image.
The table below (Table 1.0) contains all the fields that must be configured when setting up a SIP trunk in ICMS.
The majority of required fields have default configurations and don't need to manually configure unless otherwise
required for your deployment.

Table 1.0

Field Parameter Description

Outbound Default From the drop-down select the Default.
PBX Type
Outbound 10.208.14.98 IP Address of the Audio Codes SBC server
Address
Outbound 5080 This must be the port listening on SBC end.
Port
Priority 1 This is configurable based on the requirement.

SIP Trunk SIP Trunk This can be defined under the PBX Policies. You can define a new policy if required
Policy based on the requirement.

Policies > New >SIP Trunk

TYPE=SIP Trunk
Keepalive =Enabled
Protocol = UDP/TCP
Allowed Outbound Call Types= Voice services
Step 2: Configuring Voice Services

• Enabling HOOT Voice services

Navigate to Voice Services >New> Hoot, as shown in the image.

Navigate to Voice Services > Hoot > Provisioning, as shown in the image
The table below (Table 1.1) contains all the fields that must be configured for the Hoot voice service.

Table 1.1

Field Parameter Description

Call Region From the drop-down select the Required call region.
Primary SIP A.B.B.C.D From the drop-down menu select the sip trunk which is created
Trunk 10.208.14.98 with the SBC
Secondary None This shall remain with the none
SIP trunk
SIP Trunk Outbound / If you select outbound the ICS will automatically originate an
Connection Inbound outbound call to the configured SIP trunk. If you select Inbound ICS
mode will accept an inbound call. This can be configured as per the client's
requirement.
Outbound 100413700033 The number which is configured at the client end.
Address

Inbound 100513700033 Number to make an inbound call with the ICS.

Address

Navigate to Voice Services > Hoot >User permission, as shown in the image to assign it to the required user.
 • Enabling ARD Voice services

Navigate to Voice Services > New> ARD, as shown in the image.

Navigate to Voice Services > ARD > Provisioning, as shown in the image
The table below (Table 1.2) contains all the fields that must be configured for the ARD voice service.

Table 1.2

Field Parameter Description

Call Region From the drop-down select the Required call region.
Primary SIP A.B.B.C.D From the drop-down menu select the sip trunk which is created
Trunk 10.208.14.98 with the SBC
Secondary None This shall remain with the none
SIP trunk
Outbound 100413300032 The number which is configured at the client end.
Address

Inbound 100513700032 Number to make an inbound call with the ICS.

Address

Navigate to Voice Services > ARD >User permission, as shown in the image to assign it to the required user.

Step 3: Audio Codes SBC Configuration

1)Configuring SRD (Signalling Routing Domain)

The SRD is a logical representation of an entire SIP-based VoIP network (Layer 5) consisting of groups of SIP servers.
The SRD is associated with all the configuration entities (e.g., SIP Interfaces and IP Groups) required for routing calls
within the network. Typically, only a single SRD is required (recommended) for most deployments.
Multiple SRDs are only required for multi-tenant deployments, where the physical device is "split" into multiple logical
devices.

If your deployment requires only one SRD, you can use the default SRD instead of creating a new one. When only one
SRD is employed and you create other related configuration entities (e.g., SIP Interfaces), the default SRD is
automatically assigned to the new configuration entity. Therefore, when employing a single-SRD configuration
topology, there is no need to handle SRD configuration (i.e., transparent). SRDs are associated with the following
configuration entities:

• SIP Interface (mandatory)

• IP Group (mandatory)

• Proxy Set (mandatory)

• Admission Control rule

• Classification rule

Navigate to Signalling & Media > SRD > New/Edit as shown in the image.

1)Configuring SIP Interface.

The SIP Interface represents a Layer-3 network for the IP-based SIP entity in Audio Codes. It defines a local listening
port for SIP signalling traffic on a local, logical IP network interface. The term local implies that it's a logical port and
network interface on the device. The SIP Interface is used to receive and send SIP messages with a specific SIP entity
(IP Group). Therefore, you can create a SIP Interface for each SIP entity in the VoIP network with which your ICS servers
need to communicate. The SIP Interface is associated with the SIP entity, by assigning the SIP Interface to an SRD that
is in turn, assigned to the IP Group of the SIP entity

Navigate to Signalling & Media > SIP Interface > New/Edit as shown in the image.

External interface (Public SIP interface)

Internal SIP interface (SIP interface to communicate with the ICS)

Navigate to Signalling & Media > Media Realms > New/Edit as shown in the image.
The Media Realm defines a local UDP port range for RTP (media) traffic on any one of the device's
logical IP network interfaces. The Media Realm is used to receive and send media traffic with a specific SIP entity (IP
Group). The Media Realm can be associated with the SIP entity, by assigning the Media Realm to the IP Group of the
SIP entity, or by assigning it to the SIP Interface associated with the SIP entity.

Media Realm for the ICS.

Media Realm for Cadence Community

Navigate to Signalling & Media > Proxy sets > New/Edit as shown in the image.

A Proxy Set defines the address and transport type (e.g., UDP or TCP) of a SIP server (e.g., SIP proxy and SIP registrar
server). The Proxy Set represents the destination (address) of the IP Group configuration entity. Each Proxy Set can be
configured with up to 10 addresses configured as an IP address and/or DNS hostname (FQDN),
enabling you to implement load balancing and redundancy (Proxy Hot-Swap feature) between multiple servers.

Navigate to Signalling & Media > IP Groups > New/Edit as shown in the image.
An IP Group represents a SIP entity in the network with which the device communicates. This can
be a server (e.g., IP PBX or ITSP) or a group of users (e.g., LAN IP phones). For servers, the IP Group is typically used
to define the server's IP address by associating it with a Proxy Set

Navigate to Signalling & Media > SBC > IP to IP routing as shown in the image.
ICS to SBC routing.

Cisco Gateway to DC SCB

Cisco Gateway configuration

card type t1 0 0 ! Define card type, E1 or T1

card type t1 0 1 ! Define card type, E1 or T1
card type t1 0 2 ! Define card type, E1 or T1
card type t1 0 3 ! Define card type, E1 or T1

network-clock-participate wic 0 ! Specify network clock participation by card

network-clock-participate wic 1 ! Specify network clock participation by card
network-clock-participate wic 2 ! Specify network clock participation by card
network-clock-participate wic 3 ! Specify network clock participation by card
network-clock-select 1 T1 0/0/0 ! Configure network clock selection
network-clock-select 4 T1 0/3/0 ! Configure network clock selection

! Configure voice service global parameters

voice service pots
!
voice service voip
rtp-port range 16384 16782
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
options-ping 60
listen-port non-secure 5100
no call service stop

! Controller configuration:
controller T1 0/0/0
linecode ami
cablelength long 0db
ds0-group 0 timeslots 1 type e&m-immediate-start
ds0-group 1 timeslots 2 type e&m-immediate-start
ds0-group 2 timeslots 3 type e&m-immediate-start
ds0-group 3 timeslots 4 type e&m-immediate-start
ds0-group 4 timeslots 5 type e&m-immediate-start
ds0-group 5 timeslots 6 type e&m-immediate-start
ds0-group 6 timeslots 7 type e&m-immediate-start
ds0-group 7 timeslots 8 type e&m-immediate-start
ds0-group 8 timeslots 9 type e&m-immediate-start
ds0-group 9 timeslots 10 type e&m-immediate-start
ds0-group 10 timeslots 11 type e&m-immediate-start
ds0-group 11 timeslots 12 type e&m-immediate-start
ds0-group 12 timeslots 13 type e&m-immediate-start
ds0-group 13 timeslots 14 type e&m-immediate-start
ds0-group 14 timeslots 15 type e&m-immediate-start
ds0-group 15 timeslots 16 type e&m-immediate-start
ds0-group 16 timeslots 17 type e&m-immediate-start
ds0-group 17 timeslots 18 type e&m-immediate-start
ds0-group 18 timeslots 19 type e&m-immediate-start
ds0-group 19 timeslots 20 type e&m-immediate-start
ds0-group 20 timeslots 21 type e&m-immediate-start
ds0-group 21 timeslots 22 type e&m-immediate-start
ds0-group 22 timeslots 23 type e&m-immediate-start
ds0-group 23 timeslots 24 type e&m-immediate-start
!

! Voice port configuration for ARD:

voice-port 0/0/0:0
define Tx-bits idle 1111
define Tx-bits seize 0000
define Rx-bits idle 1111
define Rx-bits seize 0000
timeouts wait-release 1
connection plar 100513300032
!

! Voice port configuration for Hoot:

voice-port 0/0/0:1
timeouts call-disconnect 3
timeouts wait-release 1
connection trunk 100513300033
!

! VoIP Dial Peer to SBC

dial-peer voice 1003 voip
destination-pattern 1005........
session protocol sipv2
session target ipv4:10.143.5.4:5100
voice-class sip options-ping 60
voice-class sip options-keepalive down-interval 60 retry 3
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
codec g711ulaw
!

!POTS Dial peer for ARD

dial-peer voice 13300032 pots
destination-pattern 100413300032
port 0/0/0:0
!

!POTS Dial peer for Hoot

dial-peer voice 13300033 pots
destination-pattern 100413300033
port 0/0/0:1

Test Cases
Test Case Pass/Fail Observation Comments
DMVS voice services calls- ARD
A1 Make ARD, A1 Pass A1 make the call.
Release the Call A1 can rear the ring back.
B1 receives signalling.
Two audios established.
A1 release the call

A1 Make ARD and Pass A1 make the call.

B1 release the call A1 can hear the ring back.
B1 receives signalling.
Two way audio
established
B1 release the call

A1 Make an ARD, A2 Pass A1 make the call.

Barge into the call, A1 can rear the ring back
B1 Release B1 receives signalling
Two way audio
established
A2 barge in
B1 release
A1 and A2 stay on the call
A1 release
A2 drops

A1 Make a ARD, A2 Pass A1 make the call

barge in to the call, A1 can rear the ring back
and A2 barge out, B1 B1 receives signalling
Release Two way audio
established
A2 barge in
A2 barge out
A1 and B1 stay on the call
B1 release
A1 drops

A1 Make an ARD, A2 Pass A1 make the call

barge in to the call A1 can rear the ring back
and A1 barge out B1 receives signalling
B1 Release Two way audio
established
A2 barge in
A1 barge out
A2 and B1 stay on the call
B1 release
A2 drops
A2 making an ARD, pass A2 make the call
A1 barge in, B1 A2 can rear the ring back
Release B1 receives signalling
Two way audio
established
A1 barge in
A1 barge out
A2 and B1 stay on the call
B1 release
A2 drops

A2 Making an ARD, pass A2 make the call

A1 barge in A2 barge A2 can rear the ring back
out, B1 Release B1 receives signalling
Two way audio
established
A1 barge in
A2 barge out
A1 and B1 stay on the call
B1 release
A1 drops

A2 making an ARD pass A2 make the call.

B1 Release A2 can rear the ring back
B1 receives signalling.
Two way audio
established
B1 release the call

B1 making an ARD, pass B1 make the call.

A1 Answering B1 can rear the ring back
A1 receives signalling.
Two way audio
established
A1 release the call

B1 making an ARD, pass B1make the call.

A1 Answering, A2 B1 can rear the ring back
barge in, A1 barge A1 receives signalling.
out, A2 Release Two way audio
established
A2 barge in
A1 barge out
A2 release the call
 B1 making an ARD, pass B1make the call.
A2 Answering, A1 B1 can rear the ring back
barge in, A2 barge A2 receives signalling.
out, A1 Releasing Two way audio
established
A1 barge in
A2 barge out
A1 release the call

A1 making an ARD pass A1 making the call

with B1 and A1 put A1 can hear the ring back
B1 on hold B1 receives signalling
Audio established two
way
A1 put B1 on hold
B1 can hear the beep
A1 restore the call
Two way media
established
A1 Release the call

A1 making an ARD pass A1 making the call

with B1 and A1 place A1 can hear the ring back
the call on hold, A2 B1 receives signalling
barge in, A2 release Audio established two
the call way.
A1 put B1 on hold
B1 can hear the beep
A2 barge in and takes
control of the call
Two audio established
A2 release the call.

DMVS voice services calls- HOOT

Initiated the pass Restarted the ICS service
and taken a fresh log from
HOOT service both SBC and ICS.

Can see ICS is initiating

the HOOT service.

A1 Make HOOT pass A1 making the HOOT.

Two way audio
established
Clear the HOOT
successfully
Re-joined the HOOT
successfully
A2 Make a HOOT pass
A2 making the HOOT.
Two way audio
established
Clear the HOOT
successfully
Re-joined the HOOT
successfully

A1 Make a HOOT, A2 pass Two way audio

Barge into the call established
and barge out A2 barge in successfully
two way audio
established
A2 release the call
successfully
A1 Release the call
successfully

A2 Make a HOOT, A1 pass Two-way audio

barge into the call established
and barge out A1 Barge in
A1 successfully released
the call
A2 release the call