Avaya Polycom SpectraLink 8400 Series s Suplemento

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Avaya Solution & Interoperability Test Lab

Application Notes for Polycom® SpectraLink® 8400 Series

Telephones and Avaya Aura® Communication Manager

and Avaya Aura® SIP Enablement Services –Issue 1.0

Abstract

These Application Notes describe the procedures for configuring Polycom® SpectraLink®

8400 Series Telephones which were compliance tested with Avaya Aura® Communication

Manager and Avaya Aura® SIP Enablement Services.

The overall objective of the interoperability compliance testing is to verify Polycom®

SpectraLink® 8400 Series Telephones functionalities in an environment comprised of Avaya

Aura® Communication Manager, Avaya Aura® SIP Enablement Services, various Avaya

H.323, SIP IP Telephones, and DCP telephones.

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 procedures for configuring Polycom® SpectraLink® 8400

Series Telephones (8440 and 8450) which were compliance tested with Avaya Aura®

Communication Manager and Avaya Aura® SIP Enablement Services.

Polycom® SpectraLink® 8400 series Telephones (herein referred to as SpectraLink 8400 Series)

improve productivity and responsiveness for on-site mobile professionals across a wide range of

industries, including healthcare, retail, manufacturing and hospitality. Built on open standards,

SpectraLink 8400 Series transforms the delivery of mobile enterprise applications by bringing

the power of thin client and browser technology to front-line professionals in an easy-to-use and

easy-to-manage interface. Additionally, SpectraLink 8400 Series supports a broad range of

interfaces to enterprise-grade PBX, wireless LAN, and infrastructures to deliver maximum

interoperability with the low cost of ownership.

These Application Notes assume that Communication Manager and SIP Enablement Services are

already installed and basic configuration steps have been performed. Only steps relevant to this

compliance test will be described in this document. For further details on configuration steps not

covered in this document, consult reference [1], [2], and [3].

2. General Test Approach and Test Results

The general test approach was to place calls to and from SpectraLink 8400 Series and exercise

basic telephone operations. The main objectives were to verify the following:

Registration

Codecs (G.711MU and G.729A)

Inbound calls

Outbound calls

Hold/Resume

Call termination (origination/destination)

Transfer with Shuffling enabled (origination/destination/ attended/unattended)

Transfer with Shuffling disabled (origination/destination/ attended/unattended)

Three party conference (origination/destination)

Avaya Feature Name Extension (FNE)

oCall Park

oCall Pickup

oCall Forward (Unconditional, Busy/no answer)

MWI

Voicemail

Serviceability

2.1. Interoperability Compliance Testing

The interoperability compliance test included features and serviceability. The focus of the

interoperability compliance testing was primarily on verifying call establishment on SpectraLink

8400 Series. SpectraLink 8400 Series operations such as inbound calls, outbound calls,

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hold/resume, transfer, conference, Feature Name Extension (FNE), and SpectraLink 8400 Series

interactions with SIP Enablement Services, Communication Manager, and Avaya SIP, H.323,

and digital telephones were verified. The serviceability testing introduced failure scenarios to

see if SpectraLink 8400 Series can recover from failures.

2.2. Test Results

The test objectives were verified. For serviceability testing, SpectraLink 8400 Series operated

properly after recovering from failures such as cable disconnects, and resets of SpectraLink 8400

Series and SIP Enablement Services. SpectraLink 8400 Series successfully negotiated the codec

that was used. The features tested worked as expected.

2.3. Support

Technical support on SpectraLink 8400 Series can be obtained through the following:

Phone: (978) 292-5000, and select Option 3.

Web: http://www.polycom.com/support/index.html

3. Reference Configuration

Figure 1 illustrates a sample configuration consisting of an Avaya S8720 Servers, an Avaya

G650 Media Gateway, a SIP Enablement Services server, and SpectraLink 8400 Series. The

solution described herein is also extensible to other Avaya Media Servers and Media Gateways.

Avaya S8300D Server with an Avaya G450 Media Gateway was included in the test to provide

an inter-switch scenario. For completeness, an Avaya 4600 Series H.323 IP Telephone, Avaya

9600 Series SIP IP Telephones, Avaya 9600 Series H.323 IP Telephones, and Avaya 6400 Series

Digital Telephones, are included in Figure 1 to demonstrate calls between the SIP-based

SpectraLink 8400 Series and Avaya SIP, H.323, and digital telephones.

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Figure 1: Test Configuration of SpectraLink 8400 Series

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4. Equipment and Software Validated

The following equipment and software were used for the test configuration.

Equipment

Software/Firmware

Avaya S87020 Servers

Avaya Aura® Communication

Manager 5.2.1 (R015x.02.1.016.4)

Avaya G650 Media Gateway

TN2312BP IP Server Interface

TN799DP C-LAN Interface

TN2302AP IP Media Processor

HW11 FW044

HW01 FW028

HW20 FW118

Avaya Aura® SIP Enablement Services

SES-5.2.1.0-016.4

Avaya S8300D Media Server with Avaya G450

Media Gateway

Avaya Aura® Communication

Manager 6.0.1 (R016x.00.1.510.1) with

SP2 (00.1.510.1-18860)

Avaya 4600 and 9600 Series SIP Telephones

9620 (SIP)

9630 (SIP)

9650 (SIP)

2.6.4

Avaya 4600 and 9600 Series H.323 Telephones

4625 (H.323)

9620 (H.323)

9630 (H.323)

9650 (H.323)

2.9

3.1

Avaya 6408D+ Digital Telephone

SpectraLink 8400 Series

UCS 4.0.0.10555

5. Configure the Avaya Aura® Communication Manager

This section describes the procedure for setting up a SIP trunk between Communication Manager

and SIP Enablement Services. The steps include setting up an IP codec set, an IP network region,

IP node name, a signaling group, a trunk group, and a SIP station. Before a trunk can be

configured, it is necessary to verify if there is enough capacity to setup an additional trunk. The

highlights in the following screens indicate the values used during the compliance test. Default

values may be used for all other fields.

These steps are performed from the Communication Manager System Access Terminal (SAT)

interface. SpectraLink 8400 Series and other SIP telephones are configured as off-PBX

telephones in Communication Manager.

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5.1. Capacity Verification

Enter the display system-parameters customer-options command. Verify that there are

sufficient Maximum Off-PBX Telephones –OPS licenses. If not, contact an authorized Avaya

account representative to obtain additional licenses

display system-parameters customer-options Page 1 of 11

OPTIONAL FEATURES

G3 Version: V15 Software Package: Standard

Location: 1 RFA System ID (SID): 1

Platform: 6 RFA Module ID (MID): 1

USED

Platform Maximum Ports: 44000 10273

Maximum Stations: 36000 10122

Maximum XMOBILE Stations: 0 0

Maximum Off-PBX Telephones - EC500: 50 0

Maximum Off-PBX Telephones - OPS: 100 4

Maximum Off-PBX Telephones - PBFMC: 0 0

Maximum Off-PBX Telephones - PVFMC: 0 0

Maximum Off-PBX Telephones - SCCAN: 0 0

On Page 2 of the form, verify that the number of SIP trunks supported by the system is sufficient

for the number of SIP trunks needed. If not, contact an authorized Avaya account representative

to obtain additional licenses.

display system-parameters customer-options Page 2 of 11

OPTIONAL FEATURES

IP PORT CAPACITIES USED

Maximum Administered H.323 Trunks: 100 44

Maximum Concurrently Registered IP Stations: 18000 4

Maximum Administered Remote Office Trunks: 0 0

Maximum Concurrently Registered Remote Office Stations: 0 0

Maximum Concurrently Registered IP eCons: 0 0

Max Concur Registered Unauthenticated H.323 Stations: 5 0

Maximum Video Capable H.323 Stations: 5 0

Maximum Video Capable IP Softphones: 5 0

Maximum Administered SIP Trunks: 100 50

Maximum Administered Ad-hoc Video Conferencing Ports: 0 0

Maximum Number of DS1 Boards with Echo Cancellation: 0 0

Maximum TN2501 VAL Boards: 10 1

Maximum Media Gateway VAL Sources: 0 0

Maximum TN2602 Boards with 80 VoIP Channels: 128 1

Maximum TN2602 Boards with 320 VoIP Channels: 128 0

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

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5.2. IP Codec Set

This section describes the steps for administering a codec set in Communication Manager. This

codec set is used in the IP network region for communications between Communication

Manager and SIP Enablement Services. Enter the change ip-codec-set <c> command, where c

is a number between 1 and 7, inclusive. IP codec sets are used in Section 5.3 for configuring IP

network region to specify which codec sets may be used within and between network regions.

For the compliance testing, G.711MU, G.729A were tested for verification.

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

To configure a specific codec for Avaya 9600 Series SIP phones, the 46xxsettings.txt file must

be configured. The following shows the CODEC SETTINGS section in the 46xxsettings.txt

file that needs to be modified.

##################### CODEC SETTINGS #####################

## G.711a Codec Enabled

## Determines whether G.711 a-law codec is available on

## the phone.

## 0 for No

## 1 for Yes

## SET ENABLE_G711A 1 (This shows the default)

##Added the following statement:

SET ENABLE_G711A 0

## G.711u Codec Enabled

## Determines whether G.711 mu-law codec is available on

## the phone.

## 0 for No

## 1 for Yes

## SET ENABLE_G711U 1

## G.729 Codec Enabled

## Determines whether G.729 codec is available on the

## phone.

## 0 for G.729(A) disabled

## 1 for G.729(A) enabled without Annex B support

## 2 for G.729(A) enabled with Annex B support

## SET ENABLE_G729 1

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## G.726 Codec Enabled

## Determines whether G.726 codec is available on the

## phone. This parameter is not supported on 16cc phones.

## 0 for No

## 1 for Yes

## SET ENABLE_G726 1

## G.726 Payload Type

## Specifies the RTP payload type to be used with the

## G.726 codec. (96-127). This parameter is not supported

## on 16cc phones.

## SET G726_PAYLOAD_TYPE 110

## G.722 Codec Enabled

## Determines whether G.722 codec is available on the

## phone. This parameter is not supported on 16cc phones.

## 0 for No

## 1 for Yes

## SET ENABLE_G722 0

SET ENABLE_G722 1

## DTMF Payload Type

## Specifies the RTP payload type to be used for RFC

## 2833 signaling. (96-127).

## SET DTMF_PAYLOAD_TYPE 120

## DTMF Transmission Method

## Specifies whether DTMF tones are sent in-band, as

## regular audio, or out-of-band, using RFC 2833

## procedures.

## 1 for in-band

## 2 for out-of-band using RFC 2833

## SET SEND_DTMF_TYPE 2

5.3. Configure IP Network Region

This section describes the steps for administering an IP network region in Communication

Manager for communication between Communication Manager and SIP Enablement Services.

Enter the change ip-network-region <n> command, where n is a number between 1 and 250

inclusive, and configure the following:

Authoritative Domain –Enter the appropriate name for the Authoritative Domain. Set

to the appropriate domain. During the compliance test, the authoritative domain is set to

avaya.com. This should match the SIP Domain value on SIP Enablement Services, in

Section 6.1.

Intra-region IP-IP Direct Audio –Set to yes to allow direct IP-to-IP audio connectivity

between endpoints registered to Communication Manager or SIP Enablement Services in

the same IP network region. The default value for this field is yes.

Codec Set –Set the codec set number as provisioned in Section 5.2.

Inter-region IP-IP Direct Audio –Set to yes to allow direct IP-to-IP audio connectivity

between endpoints registered to Communication Manager or SIP Enablement Services in

different IP network regions. The default value for this field is yes.

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change ip-network-region 1 Page 1 of 19

IP NETWORK REGION

Region: 1

Location: 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? n

UDP Port Max: 3029

DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y

Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS

Audio PHB Value: 46 Use Default Server Parameters? y

Video PHB Value: 26

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 6

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

5.4. Configure IP Node Name

This section describes the steps for setting IP node name for SIP Enablement Services in

Communication Manager. Enter the change node-names ip command, and add a node name for

SIP Enablement Services along with its IP address.

change node-names ip Page 1 of 2

IP NODE NAMES

Name IP Address

ASM 10.64.40.42

CLAN 10.64.40.24

CLAN-AES 10.64.40.25

G450 10.64.41.21

MEDPRO 10.64.40.26

MM-MAS 10.64.20.63

S8300 10.64.42.21

SES 10.64.40.41

5.5. Configure SIP Signaling

This section describes the steps for administering a signaling group in Communication Manager

for communication between Communication Manager and SIP Enablement Services. Enter the

add signaling-group <s> command, where s is an available signaling group and configure the

following:

Group Type –Set to sip.

Near-end Node Name - Set to CLAN as displayed in Section 5.4.

Far-end Node Name - Set to the SIP Enablement Services name configured in Section

5.4.

Far-end Network Region - Set to the region configured in Section 5.3.

Far-end Domain - Set to avaya.com. This should match the SIP Domain value in

Section 6.1.

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Direct IP-IP Audio Connections –Set to y, since Media Shuffling is enabled during the

compliance test

add signaling-group 201 Page 1 of 1

SIGNALING GROUP

Group Number: 201 Group Type: sip

Transport Method: tls

IMS Enabled? n

Near-end Node Name: CLAN Far-end Node Name: SES

Near-end Listen Port: 5061 Far-end Listen Port: 5061

Far-end Network Region: 1

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 Direct IP-IP Early Media? n

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

5.6. Configure SIP Trunk

This section describes the steps for administering a trunk group in Communication Manager for

communication between Communication Manager and SIP Enablement Services. Enter the add

trunk-group <t> command, where t is an unallocated trunk group and configure the following:

Group Type –Set the Group Type field to sip.

Group Name –Enter a descriptive name.

TAC (Trunk Access Code) –Set to any available trunk access code.

Signaling Group –Set to the Group Number field value configured in Section 5.5.

Number of Members –Allowed value is between 0 and 255. Set to a value large enough

to accommodate the number of SIP telephone extensions being used.

Note: Each SIP call between two SIP endpoints (whether internal or external) requires two SIP

trunks for the duration of the call. The license file installed on the system controls the maximum

permitted.

add trunk-group 201 Page 1 of 21

TRUNK GROUP

Group Number: 201 Group Type: sip CDR Reports: y

Group Name: To SES COR: 1 TN: 1 TAC: 116

Direction: two-way Outgoing Display? y

Dial Access? n Night Service:

Queue Length: 0

Service Type: tie Auth Code? n

Signaling Group: 201

Number of Members: 10

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