Avaya S8300 Series Bedienungsanleitung

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SPOC 8/10/2007 Solution & Interoperability Test Lab Application Notes

ACM-Cisco-CM-GK.doc

Avaya Solution & Interoperability Test Lab

Configuring Avaya Communication Manager, Cisco 2811

Gatekeeper and Cisco CallManager to Control VoIP Calls

between Avaya and Cisco Telephones - Issue 1.0

Abstract

These Application Notes describe a configuration, including, Avaya Communication Manager,

a Cisco 2811 Gatekeeper, and Cisco CallManager that was verified for a customer proof of

concept request. In this configuration, Avaya Communication Manager and Cisco

CallManager are registered with the Cisco 2811 Gatekeeper, which provides call control

between Avaya Communication Manager and Cisco CallManager.

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SPOC 8/10/2007 Solution & Interoperability Test Lab Application Notes

ACM-Cisco-CM-GK.doc

1. Introduction

An H.323 gatekeeper can provide address translation and network access control services for

H.323 terminals and gateways. These Application Notes present the configuration steps to allow

Avaya Communication Manager to communicate with Cisco CallManager via a Cisco 2811

Gatekeeper. These Application Notes will cover the necessary configuration steps for Avaya

Communication Manager, a Cisco 2811 Gatekeeper and Cisco CallManager.

Figure 1 depicts the network configuration utilized for these Application Notes.

In Location A in Figure 1, the Avaya G350 Media Gateway with S8300 Server provides both

H.323 Gateway (GW) and Gatekeeper (GK) functionality. In Location B, the Cisco 2811 access

router is configured as an H.323 Gatekeeper, and the Cisco CallManager is registered with it.

The Cisco 2811 GK provides E.164 to IP address resolution and call control for bandwidth

between the Location A and Location B.

The are two zones defined on this Gatekeeper, a local zone for Location B and a remote zone

for Location A. The Gatekeeper stays in Local zone. The following shows the basic flow for a

call from Location A to Location B:

•Avaya Communication Manager sends a Location Request (LRQ) message to the Cisco

2811 Gatekeeper.

•The Cisco 2811 Gatekeeper checks its dial plan and bandwidth requirement

•If the call meets the bandwidth requirement:

oThe Cisco 2811 GK responds with location confirm (LCF) to Avaya

Communication Manager.

oAvaya Communication Manager will set up the call to the Cisco CallManager

through H.225 (Q.931) signaling.

•If the bandwidth requested exceeds the bandwidth requirement, the Cisco 2811

Gatekeeper sends call Location Rejection (LRJ) to Avaya Communication Manager to

deny the call.

Similarly, this call flow can be applied for call origination in Location B to Location A.

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SPOC 8/10/2007 Solution & Interoperability Test Lab Application Notes

ACM-Cisco-CM-GK.doc

Cisco CallManager 4.1

60.1.1.5

Location A

Cisco 7970 IP Telephone

60.1.1.100

x61000

Cisco Catalyst

3750

Cisco 7960 IP Telephone

60.1.1.101

x61002

Ethernet

MPLS WAN

Avaya 9630

IP Telephone

X4555

22.1.1.100

Ethernet

Cisco 2811

GateKeeper

60.1.1.10

WXY

PQR

OPE

796

CISCOIP

PHONE

messag

es directo ri

settin g

servic e

WXY

PQR

OPE

796

CISCOIP

PHONE

messag

es directo ri

settin

servic

Zone:

Remote

Avaya 4600

IP Telephone

X4550

22.1.1.101

Avaya 2420

Digital Telephone

x4501

Zone:

Local

T1 T1

DHCP/TFTP

Server

Avaya S8300 Server with

G350 Media Gateway

IP: 22.1.1.22

Location B

Figure 1: Network Diagram for these Application Notes

2. Equipment and Software Validated

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

Network Component Software Version

Avaya Communication Manager

S8300 Server with G350 Media Gateway 3.1.2 (634)

Avaya G350 Media Gateway 26.31

Avaya 9600 IP Telephone (H.323) 1.2.1

Avaya 4620 IP Telephone (H.323) 2.7

Cisco 2811 Gatekeeper 12.4(2)T5 (IP Voice)

Cisco CallManager 4.1(2)

Cisco Catalyst 3750 Switch 8.2.1

Cisco 7970/7960 IP Telephones 7.2

Table 1 - Network Component Software Versions

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SPOC 8/10/2007 Solution & Interoperability Test Lab Application Notes

ACM-Cisco-CM-GK.doc

3. Avaya Communication Manager Configuration

3.1. Configuring Node Names on the Avaya Communication Manager

This section presents configuration steps for node name to IP address mapping on Avaya

Communication Manager. It is assumed that appropriate license and authentication files have

been installed on the S8300 Server. It is also assumed that the network IP address on the S8300

Server has been configured to 22.1.1.22/24 and the IP address on the G350 GW has been

configured to 22.1.1.23/24.

Avaya Communication Manager SAT screens can be accessed using “telnet 192.11.13.6 5023”

from a computer connected to the services port, or “telnet 22.1.1.22 5023” through the network.

Use the command change node-names ip to display the mapping of node names to IP addresses.

The node name Call-Manager is mapped to Cisco CallManager’s IP address 60.1.1.5 and the

node name Cisco-GK is mapped to the Cisco 2811 router’s IP address 60.1.1.10. The procr is

mapped to the Avaya S8300 Server’s IP Address 22.1.1.22.

change node-names ip Page 1 of 1

IP NODE NAMES

Name IP Address Name IP Address

Call-Manager 60 .1 .1 .5 . . .

Cisco-GK 60 .1 .1 .10 . . .

procr 22 .1 .1 .22 . . .

3.2. Configuring IP Network Region

In this sample configuration, the IP network region 1 is configured for Location A and the IP

network region 2 is configured for Location B.

Use the command change ip-network-region 1 to specify the IP codec set to be used for

network region 1 and the IP codec set used between network region 1 and region 2. Page 1 of 19

shows that codec set 1is used for network region 1. The fields under DIFFSERV/TOS

PARAMETERS and 802.1P/Q PARAMETERS are used for QoS. The Avaya IP telephone

will receive these QoS parameters upon registration.

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ACM-Cisco-CM-GK.doc

change ip-network-region 1 Page 1 of 19

IP NETWORK REGION

Region: 1

Location: Home Domain:

Name:

Intra-region IP-IP Direct Audio: no

AUDIO PARAMETERS Inter-region IP-IP Direct Audio: no

Codec Set: 1 IP Audio Hairpinning? n

UDP Port Min: 16384

UDP Port Max: 32767 RTCP Reporting Enabled? y

RTCP MONITOR SERVER PARAMETERS

DIFFSERV/TOS PARAMETERS Use Default Server Parameters? y

Call Control PHB Value: 34

Audio PHB Value: 46

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 3

Audio 802.1p Priority: 6 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

Page 3 of 19 shows that calls between network region 1and 2are allowed, up to the 256:Kbps.

Calls above the specified limit will not be permitted, and may be re-routed to a different path.

The bandwidth usage is based on the packet size and codec selection. Avaya supports IP Call

Admission Control via Bandwidth Limits (CAC-BL). Codec set 2is used for calls between

regions 1 and 2.

change ip-network-region 1 Page 3 of 19

Inter Network Region Connection Management

src dst codec direct Total Video Dyn

rgn rgn set WAN WAN-BW-limits WAN-BW-limits Intervening-regions CAC IGAR

1 1 1

1 2 2 y 256:Kbits 0:Kbits n

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ACM-Cisco-CM-GK.doc

Use the command change ip-network-region 2 to specify the IP codec set to be used for

network region 2 and the IP codec set used between network region 1 and region 2. Page 1 of 19

shows that codec set 2is used for network region 2.

change ip-network-region 2 Page 1 of 19

IP NETWORK REGION

Region: 2

Location: Home Domain:

Name:

Intra-region IP-IP Direct Audio: no

AUDIO PARAMETERS Inter-region IP-IP Direct Audio: no

Codec Set: 2 IP Audio Hairpinning? n

UDP Port Min: 16384

UDP Port Max: 32767 RTCP Reporting Enabled? y

RTCP MONITOR SERVER PARAMETERS

DIFFSERV/TOS PARAMETERS Use Default Server Parameters? y

Call Control PHB Value: 34

Audio PHB Value: 46

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 3

Audio 802.1p Priority: 6 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

Page 3 of 19 shows that calls between network region 2and 1are allowed, up to the 256:Kbps.

Calls above the specified limit will not be permitted, and may be re-routed to a different path.

The bandwidth usage is based on the packet size and codec selection. Avaya supports IP Call

Admission Control via Bandwidth Limits (CAC-BL). Codec set 2is used for calls between

regions 2 and 1.

change ip-network-region 2 Page 3 of 19

Inter Network Region Connection Management

src dst codec direct Total Video Dyn

rgn rgn set WAN WAN-BW-limits WAN-BW-limits Intervening-regions CAC IGAR

2 1 2 y 256:Kbits 0:Kbits n

2 2 2

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SPOC 8/10/2007 Solution & Interoperability Test Lab Application Notes

ACM-Cisco-CM-GK.doc

Use the command change ip-codec-set x(where x is the codec-set number) to configure IP

codec(s). The following screens show the codec configuration. IP-codec-set 1 uses G.711MU

and ip-codec-set 2 uses G.729.

change ip-codec-set 1 Page 1 of 1

IP Codec Set

Codec Set: 1

Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)

1: G.711MU n 2 20

change ip-codec-set 2 Page 1 of 1

IP Codec Set

Codec Set: 2

Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)

1: G.729 n 2 20

3.3. Configuring Signaling and Trunk Groups to Cisco 2811

Gatekeeper

In order to configure Avaya Communication Manager as an H.323 Gatekeeper to send LRQ to

the Cisco 2811 GK, a signaling group for outbound traffic must be configured. Note that the IP

address of the Avaya S8300 Server (procr) functions as an IP address for both H.323 GW and

GK.

Use the command add signaling-group <group number> to create a signaling group. The Near-

End Node Name and Far-end Node Name are configured to the procr and Cisco-GK

respectively. To configure the Avaya S8300 Server as a H.323 GK, LRQ required must be set to

y. Near-end Listen Port and Far-end Listen Port must be configured to 1719 (UDP port 1719 for

H.225 RAS). Direct IP-IP Audio Connections and IP Audio Hairpinning must be set to n for

successful media paths. Disabling Direct IP-IP and IP Audio hairpinning will establish the RTP

stream between the G350 Media Gateway and the Cisco IP Telephones for inter location calls.

To support out-of-band DTMF, DTMF over IP must be configured to out-of-band. To control

Codec Set selection, the Far-end Network Region is configured to 2. Note that this signaling

group is configured to be associated with the trunk group 20.

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add signaling-group 20 Page 1 of 5

SIGNALING GROUP

Group Number: 20 Group Type: h.323

Remote Office? n Max number of NCA TSC: 20

SBS? n Max number of CA TSC: 0

IP Video? n Trunk Group for NCA TSC:

Trunk Group for Channel Selection: 20

Supplementary Service Protocol: a

T303 Timer(sec): 10

Near-end Node Name: procr Far-end Node Name: Cisco-GK

Near-end Listen Port: 1719 Far-end Listen Port: 1719

Far-end Network Region: 2

LRQ Required? y Calls Share IP Signaling Connection? n

RRQ Required? n H245 Control Addr On FACility? n

Media Encryption? n Bypass If IP Threshold Exceeded? n

H.235 Annex H Required? n

DTMF over IP: out-of-band Direct IP-IP Audio Connections? n

IP Audio Hairpinning? n

Interworking Message: PROGress

DCP/Analog Bearer Capability: 3.1kHz

Use the command add trunk–group <group number> to add an IP trunk group. The following

screen shows the configuration of the IP trunk group 20. As a standard IP trunk configuration,

Group Type is configured to isdn, Carrier Medium to H.323 and Service Type to tie.

add trunk-group 20 Page 1 of 21

TRUNK GROUP

Group Number: 20 Group Type: isdn CDR Reports: y

Group Name: TO Ciso-GK COR: 1 TN: 1 TAC: 120

Direction: two-way Outgoing Display? n Carrier Medium: H.323

Dial Access? n Busy Threshold: 255 Night Service:

Queue Length: 0

Service Type: tie Auth Code? n

Member Assignment Method: manual

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Note that Codeset to Send Display must be configured to 0in order for the Cisco GW to accept

the Q.931 call setup message from the Avaya Media Gateway.

add trunk-group 20 Page 2 of 21

Group Type: isdn

TRUNK PARAMETERS

Codeset to Send Display: 0Codeset to Send National IEs: 6

Charge Advice: none

Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc

QSIG Value-Added? n

Digital Loss Group: 18

Incoming Calling Number - Delete: Insert: Format: unk-pvt

In order to support caller ID on the IP trunk (isdn type), configure Send Name and Send Calling

Number to y. In the sample configuration, the Format is configured to unk-pvt.

add trunk-group 20 Page 3 of 21

TRUNK FEATURES

ACA Assignment? n Measured: none

Internal Alert? n Maintenance Tests? y

Data Restriction? n NCA-TSC Trunk Member:

Send Name: y Send Calling Number: y

Used for DCS? n Send EMU Visitor CPN? n

Suppress # Outpulsing? n Format: unk-pvt

UUI IE Treatment: service-provider

Replace Restricted Numbers? n

Replace Unavailable Numbers? n

Send Connected Number: n

Hold/Unhold Notifications? n

Send UUI IE? y Modify Tandem Calling Number? n

Send UCID? n

Send Codeset 6/7 LAI IE? y

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SPOC 8/10/2007 Solution & Interoperability Test Lab Application Notes

ACM-Cisco-CM-GK.doc

IP trunk member port values are initially configured to “ip” in the Port field. The system will

then assign a value to the port as shown below. Note the IP trunk members are virtual ports. In

the following screen, five members associated with signaling group 20 are configured. This

means that the IP trunk group only supports five simultaneous VoIP calls. The sixth call will be

blocked if there is no backup trunk available. If a PSTN trunk is configured as a backup trunk,

the additional calls can be overflowed to the PSTN, or all calls can use the PSTN if the VoIP

trunk is out of service.

display trunk-group 20 Page 5 of 21

TRUNK GROUP

Administered Members (min/max): 1/5

GROUP MEMBER ASSIGNMENTS Total Administered Members: 5

Port Name Night Sig Grp

1: T00086 20

2: T00087 20

3: T00088 20

4: T00089 20

5: T00090 20

When a call is placed from Avaya Communication Manager, the called number is routed (by

appropriate routing, e.g., UDP, AAR/ARS, etc.) to the Cisco 2811 Gatekeeper via trunk group

20. The Cisco 2811 Gatekeeper will send the call setup message to the appropriate gateway, for

these Application Notes, Cisco CallManager.

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