Avaya ClearOne Converge Pro VH20 Beilage

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ClearOneVH20-SM

Avaya Solution & Interoperability Test Lab

Application Notes for ClearOne Converge Pro VH20 with

Avaya Aura® Session Manager and Avaya Aura®

Communication Manager - Issue 1.0

Abstract

These Application Notes describe the steps required to integrate the ClearOne Converge Pro

VH20 Conferencing Solution with Avaya Aura® Session Manager and Avaya Aura®

Communication Manager using a SIP interface. The Converge Pro VH20 links other

Converge Pro products to create a complete audio conferencing system that is integrated with

an Avaya SIP telephony network. In this compliance test, Converge Pro VH20 provided SIP

connectivity for the Converge Pro 880T equipped with a speaker and microphone. The focus

of these Application Notes is on the interoperability between Converge Pro VH20, Session

Manager, and Communication Manager.

Information in these Application Notes has been obtained through DevConnect compliance

testing and additional technical discussions. Testing was conducted via the DevConnect

Program at the Avaya Solution and Interoperability Test Lab.

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1. Introduction

These Application Notes describe the steps required to integrate the ClearOne Converge Pro

VH20 Conferencing Solution with Avaya Aura® Session Manager and Avaya Aura®

Communication Manager using a SIP interface. The Converge Pro VH20 links other Converge

Pro products to create a complete audio conferencing system that is integrated with an Avaya

SIP telephony network. In this compliance test, Converge Pro VH20 provided SIP connectivity

for the Converge Pro 880T equipped with a speaker and microphone. The focus of these

Application Notes is on the interoperability between Converge Pro VH20, Session Manager, and

Communication Manager.

2. General Test Approach and Test Results

To verify interoperability of ClearOne Converge Pro VH20, linking Converge Pro 880T, with

Communication Manager and Session Manager, voice calls were made to Avaya IP telephones

(SIP and H.323). All test cases were performed manually.

2.1 Interoperability Compliance Testing

Interoperability compliance testing covered the following features and functionality:

Successful registration of Converge Pro VH20 with Session Manager.

Voice calls between Converge Pro VH20 and Avaya IP telephones (SIP and H.323),

including proper call disconnect.

Long duration calls.

G.711 and G.729 codec support and code negotiation.

Audio mute on Converge Pro VH20 and Avaya endpoints.

Converge Pro VH20 DTMF support.

Proper handling of unsuccessful calls due to call abort, no answer, dialing invalid

number, and called party busy.

Proper system recovery after a restart of Converge Pro VH20 and loss of IP connectivity.

2.2 Test Results

All test cases passed. One observation is that the Communication Manager Preferred Minimum

Session Refresh Interval setting in the SIP trunk group form and the Converge Pro VH20 Min-

SE Timer have to match, or incoming calls to VH20 will fail with a SIP Status message

indicating “Session Interval Too Small”.

2.3 Support

For technical support and information on Converge Pro VH20, contact ClearOne at:

Phone: 800-283-5936 (toll free)

Email: [email protected]

Website: http://www.clearone.com/support

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ClearOneVH20-SM

3. Reference Configuration

Figure 1 illustrates a sample configuration with an Avaya SIP-based network that includes the

following Avaya products:

Avaya Aura® Communication Manager running on an Avaya S8800 Server with a G650

Media Gateway. Communication Manager was configured as an Evolution Server.

Avaya Aura® Session Manager connected to Communication Manager via a SIP trunk

and acting as a Registrar/Proxy for SIP telephones and video endpoints.

Avaya Aura® System Manager used to configure Session Manager.

In addition, ClearOne Converge Pro VH20 registered as a SIP endpoint to Session Manager.

Converge Pro VH20 then connected to Converge Pro 880T, which provided connectivity to a

speaker and tabletop microphone. The Converge Console was used to configure the Converge

Pro VH20. All SIP devices registered with Session Manager and were configured as Off-PBX

Stations (OPS) on Communication Manager.

Note: The focus of these Application Notes is on the Converge Pro VH20.

Avaya Aura® Session Manager

(10.32.24.235)

LAN

Avaya G650 Media Gateway

Avaya Aura® System Manager

Avaya S8800 Server running

Avaya Aura® Communication Manager

Avaya 9600 Series SIP

and H. 323 IP Telephones

ClearOne Converge Console

ClearOne Converge Pro VH20

ClearOne Converge Pro 880T

Speaker Tabletop Mic

Figure 1: Avaya SIP Network with ClearOne Converge Pro VH20

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4. Equipment and Software Validated

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

Hardware Component

Version

Avaya S8800 Servers and G650 Media

Gateway

Avaya Aura® Communication Manager

6.0.1 SP 3

Avaya Aura® Session Manager

6.1 (6.1.2.0-612004)

Avaya Aura® System Manager

6.1.0 (6.1.0.4.5072-6.1.4.113)

Avaya 9600 Series IP Telephones

3.011b (H.323)

2.6 (SIP)

ClearOne Converge Pro VH20

3.0.6.15

ClearOne Converge Pro 880T

3.0.6.15

ClearOne Converge Console

3.0.5

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ClearOneVH20-SM

5. Configure Communication Manager

This section provides the procedures for configuring Communication Manager. The procedures

include the following areas:

Verify Communication Manager license

Configure Converge Pro VH20 as an Off-PBX Station (OPS)

Configure a SIP trunk between Communication Manager and Session Manager

Use the System Access Terminal (SAT) to configure Communication Manager and log in with

the appropriate credentials.

5.1 Verify OPS and SIP Trunk Capacity

Using the SAT, verify that the Off-PBX Telephones (OPS), video capable endpoints, and SIP

Trunk options are enabled on the system-parameters customer-options form. The license file

installed on the system controls these options. If a required feature is not enabled, contact an

authorized Avaya sales representative.

On Page 1, verify that the number of OPS stations allowed in the system is sufficient for the

number of SIP endpoints that will be deployed.

display system-parameters customer-options Page 1 of 11

OPTIONAL FEATURES

G3 Version: V16 Software Package: Enterprise

Location: 2 System ID (SID): 1

Platform: 28 Module ID (MID): 1

USED

Platform Maximum Ports: 65000 161

Maximum Stations: 41000 78

Maximum XMOBILE Stations: 41000 0

Maximum Off-PBX Telephones - EC500: 36000 0

Maximum Off-PBX Telephones - OPS: 41000 8

Maximum Off-PBX Telephones - PBFMC: 36000 0

Maximum Off-PBX Telephones - PVFMC: 36000 0

Maximum Off-PBX Telephones - SCCAN: 0 0

Maximum Survivable Processors: 313 0

(NOTE: You must logoff & login to effect the permission changes.)

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On Page 2 of the system-parameters customer-options form, verify that the number SIP trunks

supported by the system is sufficient.

display system-parameters customer-options Page 2 of 11

OPTIONAL FEATURES

IP PORT CAPACITIES USED

Maximum Administered H.323 Trunks: 12000 30

Maximum Concurrently Registered IP Stations: 18000 20

Maximum Administered Remote Office Trunks: 12000 0

Maximum Concurrently Registered Remote Office Stations: 18000 0

Maximum Concurrently Registered IP eCons: 414 0

Max Concur Registered Unauthenticated H.323 Stations: 100 0

Maximum Video Capable Stations: 18000 1

Maximum Video Capable IP Softphones: 18000 4

Maximum Administered SIP Trunks: 24000 30

Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0

Maximum Number of DS1 Boards with Echo Cancellation: 522 0

Maximum TN2501 VAL Boards: 128 1

Maximum Media Gateway VAL Sources: 250 0

Maximum TN2602 Boards with 80 VoIP Channels: 128 0

Maximum TN2602 Boards with 320 VoIP Channels: 128 0

Maximum Number of Expanded Meet-me Conference Ports: 300 0

(NOTE: You must logoff & login to effect the permission changes.)

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5.2 Configure SIP Trunk

In the IP Node Names form, assign an IP address and host name for the C-LAN board in the

G650 Media Gateway and the Session Manager SIP interface. The host names will be used

throughout the other configuration screens of Communication Manager.

change node-names ip Page 1 of 2

IP NODE NAMES

Name IP Address

Gateway001 10.32.24.1

ModMsg 192.50.10.45

clancrm 10.32.24.20

default 0.0.0.0

devcon-asm 10.32.24.235

medprocrm 10.32.24.21

procr 10.32.24.10

procr6 ::

( 8 of 8 administered node-names were displayed )

Use 'list node-names' command to see all the administered node-names

Use 'change node-names ip xxx' to change a node-name 'xxx' or add a node-name

In the IP Network Region form, the Authoritative Domain field is configured to match the

domain name configured on Session Manager. In this configuration, the domain name is

avaya.com. By default, IP-IP Direct Audio (shuffling) is enabled to allow audio traffic to be

sent directly between IP endpoints without using media resources in the Avaya G650 Media

Gateway. The IP Network Region form also specifies the IP Codec Set to be used for calls

routed over the SIP trunk to Session Manager. This codec set is used when its corresponding

network region (i.e., IP Network Region „1‟) is specified in the SIP signaling group.

change ip-network-region 1 Page 1 of 20

IP NETWORK REGION

Region: 1

Location: 1 Authoritative Domain: avaya.com

Name:

MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes

Codec Set: 1 Inter-region IP-IP Direct Audio: yes

UDP Port Min: 2048 IP Audio Hairpinning? y

UDP Port Max: 65535

DIFFSERV/TOS PARAMETERS

Call Control PHB Value: 34

Audio PHB Value: 46

Video PHB Value: 26

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 7

Audio 802.1p Priority: 6

Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS

H.323 IP ENDPOINTS RSVP Enabled? n

H.323 Link Bounce Recovery? y

Idle Traffic Interval (sec): 20

Keep-Alive Interval (sec): 5

Keep-Alive Count: 5

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In the IP Codec Set form, select the audio codec type supported for calls routed over the SIP

trunk to Converge Pro VH20. The form is accessed via the change ip-codec-set 1 command.

Note that IP codec set „1‟ was specified in IP Network Region „1‟ shown above. The default

settings of the IP Codec Set form are shown below. Testing was also performed with the

G.729B codec.

change ip-codec-set 1 Page 1 of 2

IP Codec Set

Codec Set: 1

Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)

1: G.711MU n 2 20

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Prior to configuring a SIP trunk group for communication with Session Manager, a SIP signaling

group must be configured. Configure the Signaling Group form as follows:

Set the Group Type field to sip.

Set the IMS Enabled field to n.

Set the Transport Method field to tcp.

Specify the C-LAN board and the Session Manager as the two ends of the signaling

group in the Near-end Node Name field and the Far-end Node Name field,

respectively. These field values were taken from the IP Node Names form.

Ensure that the TCP port value of 5060 is configured in the Near-end Listen Port and

the Far-end Listen Port fields.

The preferred codec for the call will be selected from the IP codec set assigned to the IP

network region specified in the Far-end Network Region field.

Enter the domain name of Session Manager in the Far-end Domain field. In this

configuration, the domain name is avaya.com.

The DTMF over IP field should be set to the default value of rtp-payload.

Communication Manager supports DTMF transmission using RFC 2833.

The Direct IP-IP Audio Connections field was enabled on this form.

The default values for the other fields may be used.

add signaling-group 50 Page 1 of 1

SIGNALING GROUP

Group Number: 50 Group Type: sip

IMS Enabled? n Transport Method: tcp

Q-SIP? n SIP Enabled LSP? n

IP Video? n Enforce SIPS URI for SRTP? y

Peer Detection Enabled? y Peer Server: SM

Near-end Node Name: clancrm Far-end Node Name: devcon-asm

Near-end Listen Port: 5060 Far-end Listen Port: 5060

Far-end Network Region: 1

Far-end Secondary Node Name:

Far-end Domain: avaya.com

Bypass If IP Threshold Exceeded? n

Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n

DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y

Session Establishment Timer(min): 3 IP Audio Hairpinning? n

Enable Layer 3 Test? n Initial IP-IP Direct Media? n

H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

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Configure the Trunk Group form as shown below. This trunk group is used for calls to SIP

endpoints. Set the Group Type field to sip, set the Service Type field to tie, specify the

signaling group associated with this trunk group in the Signaling Group field, and specify the

Number of Members supported by this SIP trunk group. Configure the other fields in bold and

accept the default values for the remaining fields.

add trunk-group 50 Page 1 of 21

TRUNK GROUP

Group Number: 50 Group Type: sip CDR Reports: y

Group Name: To devcon-asm COR: 1 TN: 1 TAC: 1050

Direction: two-way Outgoing Display? n

Dial Access? n Night Service:

Queue Length: 0

Service Type: tie Auth Code? n

Member Assignment Method: auto

Signaling Group: 50

Number of Members: 10

On Page 2, verify that the Preferred Minimum Session Refresh Interval (sec) field matches

the configuration on Converge Pro VH20 as described in Section 7. In this compliance test,

Converge Pro VH20 was configured to match the setting on Communication Manager. That is,

the Min SE Timer on Converge Pro VH20 was set to 180 sec. Alternatively, this timer can be

changed in the trunk group to match the setting on Converge Pro VH20.

add trunk-group 50 Page 2 of 21

Group Type: sip

TRUNK PARAMETERS

Unicode Name: auto

Redirect On OPTIM Failure: 5000

SCCAN? n Digital Loss Group: 18

Preferred Minimum Session Refresh Interval(sec): 90

Disconnect Supervision - In? y Out? y

XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n

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