Avaya Polycom SpectraLink 8400 Series s Supplement

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Solution & Interoperability Test Lab Application Notes
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Polycom8400-SES
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: [phone], 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
2.6.4
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
3.1
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
2:
3:
4:
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