Application Notes for H.323 Voice over IP Trunking between Avaya

Application Notes for H.323 Voice over IP Trunking between Avaya
Avaya Solution & Interoperability Test Lab
Application Notes for H.323 Voice over IP Trunking
between Avaya Communication Manager and VoIP
Americas Nativevoip VoIP Service - Issue 1.0
Abstract
These Application Notes describe the procedure for configuring an H.323 Voice over IP
trunk between Avaya Communication Manager and the VoIP Americas Nativevoip VoIP
service to access the Public Switched Telephone Network. The compliance testing
covered a subset of IP-to-PSTN gateways in the VoIP Americas infrastructure for H.323
IP trunking. During compliance testing, telephone calls were successfully established
over the H.323 IP trunk without shuffling, between the Avaya IP telephones, Avaya
digital telephones, and analog telephones and the telephones in the Public Switched
Telephone Network. Information in these Application Notes has been obtained through
compliance testing and additional technical discussions. Testing was conducted via the
DeveloperConnection Program at the Avaya Solution and Interoperability Test Lab with
an H.323 IP trunk terminating at the VoIP Americas Network Access Point in Florida.
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1. Introduction
These Application Notes describe the procedure for configuring H.323 Voice over IP
trunk between Avaya Communication Manager and the VoIP Americas Nativevoip VoIP
service to access the Public Switched Telephone Network (PSTN). The testing covered a
subset of IP-to-PSTN gateways in the VoIP Americas infrastructure. Nativevoip VoIP
service allows new and existing IP PBX enterprise customers to peer with the PSTN in
their native VoIP protocols, such as H.323, utilizing an IP trunk in place of traditional
analog and digital trunks to connect to the PSTN.
Figure 1 shows the compliance tested network configuration, simulating an enterprise
customer site connected via an H.323 IP trunk to the VoIP Americas Nativevoip VoIP
service to access the PSTN. The enterprise site consisted of an Avaya S8700 Media
Server and an Avaya G650 Media Gateway. The enterprise site supported Avaya IP
telephones, Avaya digital telephones, analog telephones, a fax machine and a modem.
The PSTN supported analog and digital telephones, a fax machine and a modem. All the
IP addresses in the simulated enterprise site were public IP addresses. Avaya S8700
Media Server, the IPSI, CLAN and MEDPRO resources in the Avaya G650 Media
Gateway and the Avaya IP telephones were directly connected to an Internet Service
Provider network to access Nativevoip VoIP service. VoIP Americas IP-to-PSTN
gateway was located in their Network Access Point in Florida. The IP address of VoIP
Americas IP to PSTN gateway was also publicly routable. An H.323 IP trunk was
established between the Avaya Communication Manager and VoIP Americas IP-toPSTN gateway.
Shuffling, also called Direct IP-IP Audio Connections, was disabled for the H.323 trunk
in Avaya Communication Manager. With this feature enabled, the RTP audio paths of a
call over an H.323 trunk are directly established between an Avaya IP telephone and the
terminating IP endpoint, such as VoIP Americas IP-to-PSTN gateway, without using the
IP media processor (MEDPRO) in the audio path. Since shuffling was disabled, calls
between Avaya IP telephones and PSTN telephones required the resources of the Avaya
G650 Media Gateway.
Note that the configuration is also applicable to other Avaya Media Servers and Media
Gateways. The administration of the infrastructure components in VoIP Americas
network is not the focus of these Application Notes and is not described.
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PC with modem
Avaya G650 Media Gateway
Avaya 8410D Digital
Telephone
x32001
Analog Telephone
x33001
Avaya S8700 Media Server
Fax
Avaya 4620 IP Phone
x32001
Hub
H.323 IP
trunk
(without
shuffling)
Enterprise
Avaya 4620 IP Phone
x32002
VoIP Americas
nativevoip VoIP
Service
Analog Telephone
Public Switched
Telephone Network
PC with modem
Digital Telephone
Fax
Figure 1: Sample Network Configuration
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2. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configurations
provided:
Equipment
Avaya S8700 Media Server
Avaya G650 Media Gateway
• TN799DP C-LAN
• TN2312AP IPSI
• TN2302AP MedPro
• TN2224BDigital Line
• TN793 Analog Line
Avaya 4620 IP Telephones
Avaya 8410D Digital Telephones
Analog Telephones, Fax Machine and Modems
Avaya P333T-PWR Power Over Ethernet Stackable
Switch
VoIP Americas IP to PSTN Gateway for
Nativevoip VoIP service
Software/Firmware
Avaya Communication Manager 2.2
(R012x.02.0.111.4)
HW11 FW12
HW01 FW12
HW20 FW95
10
6
2.130
4.0.17
-
3. Configure VoIP Americas IP-to-PSTN gateway for
Nativevoip VoIP Service
As a service provider, VoIP Americas is responsible for configuring the relevant IP-toPSTN gateway at their Network Access Point appropriately to work with Avaya
Communication Manager at an enterprise customer site. The following list highlights the
information that VoIP Americas administered on their IP-to-PSTN gateway in the
compliance tested network configuration. The detailed configuration information is
private and is not described here.
•
•
•
•
•
•
•
Public IP address of VoIP Americas IP-to-PSTN gateway.
Public IP address of CLAN card in the Avaya G650 Media Gateway.
IP codec set – G.711 or G.729 determined by the customer’s preference.
G.711 T.38 fax with failover to pass through as G.711 fax.
G.729 fax.
Out-of-band or in-band DTMF determined by the customer’s preference.
DID numbers and dial plan for routing the calls from PSTN to the customer site.
In addition, the PSTN connected to the IP-to-PSTN gateway was validated to carry the
non-voice traffic, such as fax.
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4. Configure Avaya Communication Manager
This section presents configuration steps for the Avaya S8700 Media Server. Before
proceeding, use the command display system-parameters special-applications and page
forward to Page 4 to verify that Special Application SA8507 is enabled. SA8507 must be
enabled to achieve the interoperability documented in these Application Notes. If SA8507
is not enabled, contact your authorized Avaya sales representative.
display system-parameters special-applications
SPECIAL APPLICATIONS
(SA8481) - Replace Calling Party Number with ASAI ANI?
(SA8500) - Expanded UUI Display Information?
(SA8506) - Altura Interoperability (FIPN)?
(SA8507) - H245 Support With Other Vendors?
(SA8508) - Multiple Emergency Access Codes?
(SA8510) - NTT Mapping of ISDN Called-Party Subaddress IE?
(SA8517) - Authorization Code By COR?
(SA8518) - Automatic Callback with Called Party Queuing?
(SA8520) - Hoteling Application for IP Terminals?
(SA8558) - Increase Automatic MWI & VuStats (S8700 only)?
(SA8567) - PHS X-Station Mobility over IP?
(SA8569) - No Service Observing Tone Heard by Agent?
(SA8573) - Call xfer via ASAI on CAS Main?
(SA8582) - PSA Location and Display Enhancements?
(SA8587) - Networked PSA via QSIG Diversion?
(SA8589) - Background BSR Polling?
(SA8601) - Two-Digit AUX Reason Codes?
(SA8608) - Increase Crisis Alert Buttons (S8700 only)?
(SA8621) - SCH Feature Enhancements?
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Solution & Interoperability Test Lab Application Notes
©2005 Avaya Inc. All Rights Reserved.
Page
4 of
n
n
n
y
n
n
n
n
n
n
n
n
n
n
n
n
n
n
n
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5
4.1. Configuring VoIP Attributes
This section illustrates the parameters used in the administration of the H.323 Signaling Group and
IP Trunk Group. See Section 4.2 for their relevant usage.
4.1.1. IP Audio Codec Set
Administer the desired audio codec – G.711 or G.729 – using the IP Codec Set form. To specify
the codecs, enter change ip-codec-set p using the System Access Terminal (SAT), where p is the
number of a codec set, and modify the IP Codec Set form accordingly. The default settings are
shown below:
change ip-codec-set 1
Page
1 of
2
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2:
Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20
To enable T.38 fax routing, page forward to Page 2 and set the FAX to t.38-standard, as illustrated
in the following example. Note that T.38 fax requires the TN2302AP Media Processor with version
HW10 or greater and FW95 or greater. (For regular fax routing, page forward to Page 2 and set the
FAX to off. This will treat the fax as an ordinary voice call).
change ip-codec-set 1
Page
2 of
2
IP Codec Set
FAX
Modem
TDD/TTY
Mode
t.38-standard
off
US
Redundancy
0
0
3
For modem calls, keep the default setting of Modem as off. The modem calls will be treated as
voice calls.
4.1.2. IP Network Region
The most relevant attribute on IP Network Region form related to this application is the Codec Set.
The IP Network Region is used to obtain the codec set used for negotiation of trunk bearer
capability.
To configure the IP network region, enter change ip-network-region m using the SAT, where m is
the number of the region. On Page 1 of the IP Network Region form, modify the Codec Set to the
number of the codec set that will be used in this region. The following example illustrates that
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codec set 1 will be used for IP network region 1. Note that the Shuffling, also called Direct IP to IP
Audio Connections, is disabled on the Signaling form – see Section 4.2.2.
change ip-network-region 1
Page
1 of 19
IP NETWORK REGION
Region: 1
Location:
Name:
Home Domain:
Intra-region IP-IP Direct Audio: yes
Inter-region IP-IP Direct Audio: yes
IP Audio Hairpinning? y
AUDIO PARAMETERS
Codec Set: 1
UDP Port Min: 2048
UDP Port Max: 3049
RTCP Reporting Enabled? n
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 34
Audio PHB Value: 46
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 7
Audio 802.1p Priority: 6
H.323 IP ENDPOINTS
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
AUDIO RESOURCE RESERVATION PARAMETERS
RSVP Enabled? n
4.2. Configuring H.323 IP Trunk between Avaya Communication
Manager and VoIP Americas
This section illustrates the parameters used in the administration of the H.323 Signaling Group and
IP Trunk Group.
4.2.1. IP Node Names
The following illustrates a subset of the IP Node Names screen that maps logical names to IP
addresses. These node names are presented because they will appear in other screens, such as the
screen defining the H.323 signaling group. Note that the IP addresses for CLAN, MedPro and
VoIPAmerias in the compliance tested configuration were administered as public IP addresses (not
shown here).
Page
change node-names ip
Name
CLAN
MedPro
VoIPAmericas
default
procr
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IP NODE NAMES
Name
IP Address
0
. . .
.0
.0
.0
1 of
1
IP Address
.
.
.
.
.
.
.
.
.
.
.
.
. . .
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4.2.2. Signaling Group
Administer a signaling group by entering change signaling-group n, where n is the number of the
signaling group number.
• Group Type: Enter h.323 for Group Type.
• Node names and the listen ports: The CLAN is the near-end of the signaling group. The farend is set to VoIPAmericas (the node name of the VoIP Americas IP to PSTN gateway).
Retain the default near-end listen port (1720) and enter 1720 as the far-end listen port. In
general, the Far-end Network Region field can be left blank, or it can be populated with a
network region number. In the compliance-tested configuration, the Far-end Network Region
field is set to 1.
• Direct IP-IP Audio Connections – This field must be set to n to disable shuffling, or
interoperability problems will be experienced. Since shuffling must remain disabled for the
signaling group, calls between Avaya IP telephones and PSTN telephones will require the
media processor resources of the Avaya G650 Media Gateway.
• DTMF over IP: Set this field to either out-of-band or in-band. Note that in-band DTMF is
relevant to G.711 calls only. This setting must match the DTMF setting in VoIP Americas IP to
PSTN gateway (see Section 3).
The following example shows how to add a signaling group at the main site.
add signaling-group 1
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1 of
5
SIGNALING GROUP
Group Number: 1
Group Type: h.323
Remote Office? n
SBS? n
Trunk Group for Channel Selection:
Supplementary Service Protocol: a
T303 Timer(sec): 10
Near-end Node Name: CLAN
Near-end Listen Port: 1720
LRQ Required? n
RRQ Required? n
Media Encryption? n
DTMF over IP: out-of-band
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Max number of NCA TSC: 0
Max number of CA TSC: 0
Trunk Group for NCA TSC:
Network Call Transfer? n
Far-end Node Name: VoIPAmericas
Far-end Listen Port: 1720
Far-end Network Region: 1
Calls Share IP Signaling Connection? n
H245 Control Addr On FACility? n
Bypass If IP Threshold Exceeded? n
Direct IP-IP Audio Connections? n
IP Audio Hairpinning? n
Interworking Message: PROGress
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4.2.3. Trunk Group
Configure an H.323 IP trunk. Enter add trunk-group n, where n is the trunk group number.
Administer the trunk group parameters, with the following settings
• Group Type: Enter isdn.
• TAC: Enter a trunk access code, such as 910, according to the dial plan.
• Carrier Medium: Enter IP.
• Service Type: Enter tie to set this as an IP tie trunk between the two servers.
add trunk-group 10
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22
TRUNK GROUP
10
Group Type: isdn
CDR Reports: y
OUTSIDE CALL
COR: 1
TN: 1
TAC: 910
two-way
Outgoing Display? y
Carrier Medium: IP
y
Busy Threshold: 255
Night Service:
0
tie
Auth Code? n
TestCall ITC: rest
Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS
Codeset to Send Display: 6
Codeset to Send National IEs: 6
Max Message Size to Send: 260
Charge Advice: none
Supplementary Service Protocol: a
Digit Handling (in/out): enbloc/enbloc
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
Trunk Hunt: ascend
Digital Loss Group: 18
Incoming Calling Number - Delete:
Insert:
Format:
Bit Rate: 1200
Synchronization: async
Duplex: full
Disconnect Supervision - In? y Out? n
Answer Supervision Timeout: 0
4.2.4. Trunk Group Members
Configure the trunk group members on Page 6 of the trunk-group form, by setting the port to IP
and the previously configured signaling group. After submitting the form, the port field values are
changed as shown below.
display trunk-group 10
GROUP MEMBER ASSIGNMENTS
1:
2:
3:
4:
5:
Port
T00207
T00185
T00186
T00187
T00188
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Code Sfx Name
Page
TRUNK GROUP
Administered Members (min/max):
Total Administered Members:
Night
6 of 22
1/5
5
Sig Grp
1
1
1
1
1
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4.2.5. Associate Signaling Group to an IP trunk Group
The signaling group is associated with the IP Trunk Group. Using the command change signalinggroup 1, enter the number 10 in the Trunk Group for Channel Selection field.
change signaling-group 1
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1 of
5
SIGNALING GROUP
Group Number: 1
Group Type: h.323
Remote Office? n
SBS? n
Trunk Group for Channel Selection: 10
Supplementary Service Protocol: a
T303 Timer(sec): 10
Near-end Node Name: CLAN
Near-end Listen Port: 1720
LRQ Required? n
RRQ Required? n
Media Encryption? n
DTMF over IP: out-of-band
Max number of NCA TSC: 0
Max number of CA TSC: 0
Trunk Group for NCA TSC:
Network Call Transfer? n
Far-end Node Name: VoIPAmericas
Far-end Listen Port: 1720
Far-end Network Region: 1
Calls Share IP Signaling Connection? n
H245 Control Addr On FACility? n
Bypass If IP Threshold Exceeded? n
Direct IP-IP Audio Connections? n
IP Audio Hairpinning? n
Interworking Message: PROGress
5. Interoperability Compliance Testing
The interoperability compliance testing focused on assessing H.323 Voice over IP trunking
(without shuffling) between Avaya Communication Manager and VoIP Americas IP to PSTN
gateway to provide VoIP Americas Nativevoip VoIP service. An H.323 IP trunk was established
between the Avaya Communication Manager at a simulated enterprise site and VoIP Americas IP
to PSTN gateway in Florida. Shuffling of audio path over the H.323 trunk was disabled. Avaya
Communication Manager was configured to route inbound and outbound calls over the H.323 IP
trunk. VoIP Americas IP to PSTN gateway was configured to route the inbound calls from PSTN to
Avaya Communication Manager, and route the outbound calls from Avaya Communication
Manager to the PSTN.
5.1. General Test Approach
The general approach was to establish calls between Avaya IP telephones, Avaya digital
telephones, and analog telephones and the telephones in the Public Switched Telephone Network.
IP addresses for all the Avaya Communication Manager resources, such as IPSI, CLAN and
MEDPRO cards, Avaya IP telephones and the VoIP Americas IP to PSTN gateway were public IP
addresses. The main objectives were to verify that:
•
An H.323 IP trunk can be established between Avaya Communication Manager at a
simulated enterprise site and VoIP Americas IP to PSTN gateway in Florida, without
shuffling, to provide VoIP Americas Nativevoip VoIP service to the enterprise customers
(Nativevoip VoIP trunk).
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©2005 Avaya Inc. All Rights Reserved.
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•
•
•
•
•
•
•
Outbound voice calls from Avaya IP telephones, Avaya digital telephones, and analog
telephones via Nativevoip VoIP trunk can be routed to the telephones in the Public
Switched Telephone Network.
Inbound voice calls from telephones in Public Switched Telephone Network can be routed
to Avaya IP telephones, Avaya digital telephones, and analog telephones via Nativevoip
VoIP trunk.
Calling number is displayed on the telephones is displayed, where applicable.
Inbound and outbound voice calls can be established using G.711 and G.729 codec sets.
DTMF can be transmitted from an Avaya Communication Manager telephone towards the
PSTN and vice versa.
For G.711 and G.729 codecs, inbound and outbound fax can be transmitted using the T.38
standard protocol.
For G.711 codec, outbound modem calls can be established.
5.2. Test Results
All the tests, outlined as the main objectives in Section 5.1, completed successfully. Nativevoip
VoIP trunk, without shuffling, was successfully established. Voice, fax and modem calls and
DTMF transmission for the calls using VoIP Americas Nativevoip VoIP service established with
good quality.
6. Verification Steps
The following steps may be used to verify the configuration:
•
•
Establish, maintain, and tear down inbound and outbound voice, fax and modem calls over
the Nativevoip VoIP trunk, as outlined in Section 5.1. Verify the voice quality is good and
the DTMF are transmitted. Verify that the T.38 fax and modem calls are successful.
Make simultaneous inbound and outbound calls and verify that voice quality is acceptable.
7. Support
For technical support on the VoIP Americas Nativevoip VoIP service, contact
support@voipamericas.com or contact VoIP Americas Support at 1.305.667.3473.
8. Conclusion
As illustrated in these Application Notes, using H.323 trunks Avaya Communication Manager can
interoperate with VoIP Americas Nativevoip VoIP service to access the PSTN. For successful
interoperability with the VoIP Americas NativevoipVoIP service, the main feature settings in
Avaya Communication Manager are as follows. First, Special Application SA8507 must be
enabled. Second, Direct IP-IP audio connections, often referred to as “shuffling”, must be disabled
on the Avaya H.323 Signaling Group.
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©2005 Avaya Inc. All Rights Reserved.
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9. Additional References
The following documents are relevant to these Application Notes:
1) Administrator’s Guide for Avaya Communication Manager, January 2005, Document
Number 555-233-506.
2) Administration for Network Connectivity for Avaya Communication Manager, January
2005, Document Number 555-233-504.
Additional product documentation for Avaya products may be found at http://support.avaya.com.
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©2005 Avaya Inc. All Rights Reserved.
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©2005 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™
are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the
property of their respective owners. The information provided in these Application Notes is
subject to change without notice. The configurations, technical data, and recommendations
provided in these Application Notes are believed to be accurate and dependable, but are
presented without express or implied warranty. Users are responsible for their application of any
products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DeveloperConnection Program at devconnect@avaya.com.
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Solution & Interoperability Test Lab Application Notes
©2005 Avaya Inc. All Rights Reserved.
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