Application Notes for Configuring Direct SIP Trunking from Avaya

Application Notes for Configuring Direct SIP Trunking from Avaya
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Direct SIP Trunking
from Avaya Communication Manager using an Acme
Packet Net-Net Session Director and a SIP PSTN Gateway
– Issue 1.0
Abstract
These Application Notes describe the configuration of Direct SIP Trunking from Avaya
Communication Manager to an Acme Packet Net-Net Session Director and a SIP PSTN
gateway. The SIP PSTN gateway provided ISDN PRI trunks to a telecommunications service
provider network for PSTN interoperability. In this configuration, an Avaya Session
Enablement Services (SES) edge server is not used as part of the SIP trunking solution.
Information in these Application Notes has been obtained through DeveloperConnection
compliance testing and additional technical discussions. Testing was conducted via the
DeveloperConnection Program at the Avaya Solution and Interoperability Test Lab.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
1 of 41
cm4AcmeSipPstn
1. Introduction
These Application Notes describe the configuration of Direct SIP Trunking from Avaya
Communication Manager to an Acme Packet Net-Net Session Director and a SIP PSTN gateway.
The SIP PSTN gateway provided ISDN PRI trunks to a telecommunications service provider
network for PSTN interoperability. In this configuration, an Avaya Session Enablement Services
(SES) edge server is not used as part of the SIP trunking solution. This configuration (not
involving the SES) will be referred to in these Application Notes as “Direct SIP Trunking”.
Direct SIP Trunking uses the Acme Packet Net-Net Session Director (SD) features to distribute
SIP signaling for incoming calls to multiple Avaya Communication Manger C-LAN interfaces.
This provides for additional capacity, load balancing and survivability options (upon a C-LAN
failure or isolation event). In addition, the Acme Packet SD performs conversion between the
TCP transport for SIP messages used by the Avaya Communication Manager and the UDP
transport commonly used by other communication elements and service provider networks.
The configuration tested is shown in Figure 1. The Acme Packet SD is a session border
controller that acts as an intermediary to manage the SIP signaling and RTP media packets
between Avaya Communication Manager and other SIP terminations (such as the SIP PSTN
gateway or a SIP trunk directly to a telecommunication service provider.) The session border
controller often resides at the boundary of the enterprise customer’s IP network and serves as a
security device to isolate the customer’s internal network from the public domain. Within these
Application Notes, the networking was configured in a similar manner; the subnet used by
Avaya Communication Manager was not directly connected to the subnet used by the SIP PSTN
gateway and all SIP and RTP packets were routed through the Acme Packet SD. Other
configurations are possible, but not addressed herein.
These Application Notes specifically address the following capabilities that were verified within
the Avaya Solution and Interoperability Test Lab in Lincroft, NJ:
• Incoming PSTN calls to Avaya Communication Manager IP and Digital telephones.
• Outbound PSTN calls from Avaya (H.323) IP and Digital telephones.
• Trunk to trunk forwarding (tandem routing) of an inbound PSTN call to another PSTN
telephone via Avaya Communication Manager.
• G.711mu and G.729a codecs.
• Direct IP-IP media between (H.323) IP telephones and SIP trunks.
• Direct IP-IP media for SIP trunk to SIP trunk forwarded calls.
• Use of hunt groups, ACD splits, announcements, vectors and auto-attendant applications.
• Load balancing of incoming calls across multiple C-LAN SIP interfaces.
• Alternate routing upon failure of C-LAN SIP interface for incoming and outgoing calls.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
2 of 41
cm4AcmeSipPstn
Test Configuration for CM 4.0 Direct SIP Trunk Verification
800-222-4000
800-222-4001
800-222-5000
800-222-6000
800-222-6001
800-222-6002
800-223-6003
Avaya S8500 Media Server with
Avaya Communication Manager Release 4.0
COMPACT
Exi t
Pr ev
PAG E
L EF T
PAGE
RIG HT
SIP
PSTN
Gateway
T1
ISDN PRI
1-733-333-2225
1-733-333-2246
1-732-xxx-xxxx
142.16.57.2
SIP via UDP
RTP via UDP
Realm:
pstn_cm4
Simulated
PSTN
PSTN
- or –
SIP Service Provider
PSTN
1-732-xxx-xxxx
.201
SIP via TCP
RTP via UDP
.130 (virtual)
.131 (pri),
132 (sec)
.100
DCP Line
MEDPRO_01a05
CLAN_01a03
CLAN_01a02
CLAN_01a04
.115
.114
.113
.200
CM4 Subnet - 150.100.100.0/24
PHONE/EXIT
Me nu
Realm:
cm4_clan
142.16.58.2 (virtual),
.3 pri,
.4 sec
dcp / tdm
.112
.101
IPSI
Acme Packet
Net-Net Session Director
Version 4.1.0 p12
(high availability config )
G3si Phones
x32225
x32246
PHONE/EXIT
OPT ION S
PAG E
L EF T
PAGE
RIG HT
OPT ION S
N ex t
SPEAK ER
HO LD
SPEAKER
HOL D
6416D+M
HEADSET
Speaker
Mu te
TR AN SF ER
Tr ansf er
AB C
M UT E
Ri ng
P QRS
DEF
9
TRANSF ER
ABC
CONF ERENCE
M UT E
1
GH I
DRO P
6
W XYZ
8
3
MNO
5
T UV
7
9
2
J KL
4
6
WX YZ
8
1
GH I
3
MN O
5
TU V
0
DE F
2
JK L
4
7
H EAD SET
Hol d
A BC
R edial
1
GH I
P QR S
*
9
C ONFER EN CE
D ROP
6
WXYZ
8
3
MNO
5
TU V
7
DEF
2
J KL
4
P QRS
REDIAL
R ED IAL
#
6416 Digital Phone
x5000
Network Region 1
- Cabinet 1
- CLAN_01a02
- MEDPRO_01a05
- IP Phones
*
0
#
4621SW
IP Phone
x4000
*
0
#
4621SW
IP Phone
x4001
Network Region 2
- CLAN_01a03
- CLAN_01a04
Load Balancing
SessionGroup: cm4_sipTrkClans
150.100.100.113
150.100.100.114
RoundRobin technique
Inbound Calling (from simulated pstn - G3Si phones)
9-222-4000 to 800-222-4000 to x4000
9-222-4001 to 800-222-4001 to x4001
9-222-5000 to 800-222-5000 to x5000
9-222-6001 to 800-222-6001 to hunt group 1 x6001
9-222-6002 to 800-222-6002 to hunt group 2 (acd split) x6002
9-222-6003 to 800-222-6003 to auto-attendant vector/vdn x6003
9-223-6000 to 800-223-6000 to 1-732-555-1234 (trunk to trunk routing)
Outbound Calling (from CM4 phones)
9-1-733-333-xxxx to 1-733-333-xxxx to 3xxxx
9-1-732-xxx-xxxx to 1-732-xxx-xxxx to real pstn phones (via G3si)
Figure 1 – Direct SIP Trunking Test Configuration
2. Equipment and Software Validated
The following products and software were used for the configuration in Figure 1.
Component
Avaya
Avaya S8500B Media Server
Version
Communication Manager 4.0
(R014x.00.0.730.5)
Avaya G650 Media Gateway
TN2312BP IP Server Interface (IPSI)
HW12 FW036
TN799DP Control-LAN (C-LAN)
HW01 FW017
TN2602AP IP Media Processor (Medpro) HW02 FW025
TN2224CP Digital Line
HW08 FW015
Avaya 4621SW IP (H.323) Telephones
Release 2.2 (a20d01b2_2.bin)
Avaya 6416D+M Digital Telephone
n/a
Acme Packet
Net-Net 4000 Session Director
4.1.0 P12
(2 units in a high availability configuration)
Licensed Features: SIP, Routing, Load
Balancing, High Availability, PAC
Table 1 – Equipment and Version
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
3 of 41
cm4AcmeSipPstn
3. Configure Avaya Communication Manager
The Avaya Communication Manager was installed and configured for basic station to station
calling prior to beginning the configuration shown in these Application Notes. These basic
configuration details are outside the scope of the SIP trunking application and not included here.
3.1. SIP Trunk Configuration
3.1.1. Verify System Capacity and Required Features
The Avaya Communication Manager license controls the customer options. Contact an
authorized Avaya sales representative for assistance if insufficient capacity exists or a required
feature is not enabled.
Verify that there is sufficient remaining SIP trunk capacity available for the SIP PSTN gateway
as well as any other SIP trunking applications in use.
This is done by displaying Page 2 of the System-Parameters Customer-Options form. The
number of SIP trunks available to assign to new or existing trunk groups is the difference
between the Maximum Administered SIP Trunks and the USED value.
display system-parameters customer-options
OPTIONAL FEATURES
Page
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable H.323 Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
0
5
0
0
0
0
0
0
100
USED
0
2
0
0
0
0
0
0
20
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum Media Gateway VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
0
10
0
128
128
0
0
1
0
2
0
0
2 of
10
(NOTE: You must logoff & login to effect the permission changes.)
Figure 2: System-Parameters Customer-Options Form – Page 2
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
4 of 41
cm4AcmeSipPstn
Verify that the Automatic Route Selection (ARS) feature is enabled on Page 3 of the SystemParameters Customer-Options form.
display system-parameters customer-options
OPTIONAL FEATURES
Abbreviated Dialing Enhanced List?
Access Security Gateway (ASG)?
Analog Trunk Incoming Call ID?
A/D Grp/Sys List Dialing Start at 01?
Answer Supervision by Call Classifier?
ARS?
ARS/AAR Partitioning?
ARS/AAR Dialing without FAC?
ASAI Link Core Capabilities?
ASAI Link Plus Capabilities?
Async. Transfer Mode (ATM) PNC?
Async. Transfer Mode (ATM) Trunking?
ATM WAN Spare Processor?
ATMS?
Attendant Vectoring?
n
n
n
n
n
y
y
n
n
n
n
n
n
n
n
Page
3 of
Audible Message Waiting?
Authorization Codes?
CAS Branch?
CAS Main?
Change COR by FAC?
Computer Telephony Adjunct Links?
Cvg Of Calls Redirected Off-net?
DCS (Basic)?
DCS Call Coverage?
DCS with Rerouting?
10
n
n
n
n
n
n
n
n
n
n
Digital Loss Plan Modification? n
DS1 MSP? n
DS1 Echo Cancellation? n
(NOTE: You must logoff & login to effect the permission changes.)
Figure 3: System-Parameters Customer-Options Form – Page 3
3.1.2. Determine Node Names
Use the “change node-names ip” command to view (or assign) the node names to be used in this
configuration.
• “acme_cm4-side” and “150.100.100.130” are the Name and IP Address of the Acme
Packet SD interface where Avaya Communication Manager SIP trunk messages are
sent.
•
“clan_01a03” and “150.100.100.113” are the Name and IP Address of the TN799DP
C-LAN interface used for the first SIP signaling group (with the Acme Packet SD).
•
“clan_01a04” and “150.100.100.114” is the Name and IP Address of the C-LAN
interface used for the second SIP trunk group (with the Acme Packet SD).
change node-names ip
Page
1 of
2
IP NODE NAMES
Name
acme_cm4-side
clan_01a02
clan_01a03
clan_01a04
default
medpro_01a05
procr
val_01a08
IP Address
150.100.100.130
150.100.100.112
150.100.100.113
150.100.100.114
0.0.0.0
150.100.100.115
150.100.100.100
150.100.100.118
Figure 4: IP Node Names
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
5 of 41
cm4AcmeSipPstn
3.1.3. Define IP Codec Sets
This configuration uses two IP codec sets.
• G.711mu codec is used for local voice calls between Avaya telephones. This is IP codec
set 1.
• G.729a and G.711mu codecs (in that priority) are used for voice calls via the SIP trunks
to the SIP PSTN gateway. This is IP codec set 2.
Using “change ip-codec-set 1” command, enter “G.711MU” as the only Audio Codec. Retain
the defaults for the remaining fields.
change ip-codec-set 1
Page
1 of
2
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2:
3:
Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20
Figure 5: IP Codec Set 1
Using “change ip-codec-set 2” command, enter “G.729A” and “G.711MU” as the first and
second Audio Codec values on the form. Again, retain the defaults for the remaining fields.
change ip-codec-set 2
Page
1 of
2
IP Codec Set
Codec Set: 2
Audio
Codec
1: G.729A
2: G.711MU
3:
Silence
Suppression
n
n
Frames
Per Pkt
2
2
Packet
Size(ms)
20
20
Figure 6: IP Codec Set 2
3.1.4. Verify Near End IP Network Region
These Application Notes use IP network region 1 (the normal default) for the G650 Media
Gateway, the IP telephones and the C-LAN (in slot 1a02) used for IP telephone registration.
This will be the near-end network region for calls to the SIP PSTN gateway.
Use the “display cabinet n” command (when “n” is “1” in this case) to verify the IP Network
Region assignment of the G650-port carrier.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
6 of 41
cm4AcmeSipPstn
display cabinet 1
CABINET
CABINET DESCRIPTION
Cabinet: 1
Cabinet Layout: G650-rack-mount-stack
Cabinet Type: expansion-portnetwork
Location: 1
Rack: row6
Room: sitl
CARRIER DESCRIPTION
Carrier
Carrier Type
E
D
C
B
A
IP Network Region: 1
not-used
not-used
not-used
not-used
G650-port
Floor:
Building:
Number
PN
PN
PN
PN
PN
01
01
01
01
01
Use the “change ip-network-map” command to assign the IP telephones to Region 1. In these
Application Notes, the IP telephone addresses are with the range specified by the From IP
Address and To IP Address fields (as shown in Figure 1.)
change ip-network-map
Page
1 of
32
IP ADDRESS MAPPING
From IP Address
150.100.100.200
.
.
.
.
.
.
.
.
.
(To IP Address
150.100.100.210
.
.
.
.
.
.
.
.
.
Subnet
or Mask)
Region
1
VLAN
n
n
n
n
Emergency
Location
Extension
Figure 7: IP Network Map for IP Telephones
3.1.5. Configure the C-LAN IP Network Region Assignment
In these Application Notes, three C-LAN’s are assumed to been previously installed as part of
the initial Avaya Communication Manager basic installation (using the procedures as described
in [2]) and assigned the Node Names shown in Figure 4. The configuration in this section will
assign them to the Network Regions appropriate for Direct SIP Trunking application.
Using the “change ip-interface uucss” command (where uu is the cabinet, c is carrier, and ss is
the slot of the respective C-LAN), assign the Network Region value as follows:
• C-LAN “1a02” to Network Region 1
• C-LAN “1a03” to Network Region 2
• C-LAN “1a04” to Network Region 2
Note: In order to change an existing ip-interface, the Enable Ethernet Port must first be set to
“n”, the form saved and then the “change ip-interface uucss” done again. The Enable Ethernet
Port must then be re-enabled with “y” when the Network Region value is set.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
7 of 41
cm4AcmeSipPstn
The resulting ip-interface of the C-LAN used for IP (H.323) telephone registration is:
change ip-interface 1a02
Page
1 of
1
IP INTERFACES
Type:
Slot:
Code/Suffix:
Node Name:
IP Address:
Subnet Mask:
Gateway Address:
Enable Ethernet Port?
Network Region:
VLAN:
C-LAN
01A02
TN799 D
clan_01a02
150.100.100.112
255.255.255.0
.
.
.
y
1
n
Link: 12
Allow H.323 Endpoints? y
Allow H.248 Gateways? y
Gatekeeper Priority: 5
Target socket load and Warning level: 400
Receive Buffer TCP Window Size: 8320
ETHERNET OPTIONS
Auto? n
Speed: 100Mbps
Duplex: Full
Figure 8: IP Interface of C-LAN 1a02 used for IP Telephones
The resulting ip-interface if the C-LAN to be used for SIP signaling group 3 is:
change ip-interface 01a03
Page
1 of
1
IP INTERFACES
Type:
Slot:
Code/Suffix:
Node Name:
IP Address:
Subnet Mask:
Gateway Address:
Enable Ethernet Port?
Network Region:
VLAN:
C-LAN
01A03
TN799 D
clan_01a03
150.100.100.113
255.255.255.0
.
.
.
y
2
n
Link: 13
Allow H.323 Endpoints? y
Allow H.248 Gateways? y
Gatekeeper Priority: 5
Target socket load and Warning level: 400
Receive Buffer TCP Window Size: 8320
ETHERNET OPTIONS
Auto? n
Speed: 100Mbps
Duplex: Full
Figure 9: IP Interface of C-LAN 1a03 used for SIP Signaling Group 3
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
8 of 41
cm4AcmeSipPstn
The resulting ip-interface if the C-LAN to be used for SIP signaling group 4 is:
change ip-interface 01a04
Page
1 of
1
IP INTERFACES
Type:
Slot:
Code/Suffix:
Node Name:
IP Address:
Subnet Mask:
Gateway Address:
Enable Ethernet Port?
Network Region:
VLAN:
C-LAN
01A04
TN799 D
clan_01a04
150.100.100.114
255.255.255.0
.
.
.
y
2
n
Link: 14
Allow H.323 Endpoints? y
Allow H.248 Gateways? y
Gatekeeper Priority: 5
Target socket load and Warning level: 400
Receive Buffer TCP Window Size: 8320
ETHERNET OPTIONS
Auto? n
Speed: 100Mbps
Duplex: Full
Figure 10: IP Interface of C-LAN 1a04 used for SIP Signaling Group 4
3.1.6. Define IP Network Regions
IP network regions set various IP network properties for SIP trunk groups and other IP elements
(such as IP telephones, media processor cards, etc.) assigned to the region.
In these Application Notes, two distinct IP network regions are defined.
• “IP Network Region 1” serves as the default region for Avaya Communication Manager
and defines properties for local extension to extension calling.
• “IP Network Region 2” defines the properties for calls routed via SIP trunks to the SIP
PSTN gateway via the Acme Packet SD.
Using the “change ip-network-region 1” command, enter on Page 1:
• Name: a descriptive string such as “Avaya CM Main Location”.
• Authoritative Domain: leave blank (for ip-network-region 1).
• Codec Set: the value “1” corresponding to the ip-codec-set defined in Section 3.1.3 for
local calls between telephones on Avaya Communication Manager.
• Intra-region IP-IP Direct Audio: the value “yes” (the default).
• Inter-region IP-IP Direct Audio: the value “yes” (the default).
The IP-IP Direct Audio settings ensure the most efficient use of TN2602AP Media Processor
resources.
Defaults for the remaining values are also used.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
9 of 41
cm4AcmeSipPstn
change ip-network-region 1
Page
1 of
19
IP NETWORK REGION
Region: 1
Location:
Authoritative Domain:
Name: Avaya CM Main Location
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: 3329
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
Figure 11: IP Network Region 1 – Page 1
Page 3 of the IP network region form is used to define the codec set and connectivity
characteristics between IP network regions.
In these Application Notes, region 1 and 2 are directly connected (using the local LAN) without
bandwidth restrictions. Calls between these regions are to use codec set 2 (as defined within
Section 3.1.3).
On Page 3, configure the “src rgn 1 dst rgn 2” row as follows:
• codec set: enter “2”, to use the codec choices defined in Section 3.1.3 for calls with the
SIP PSTN gateway.
• direct WAN: enter “y” to indicate that regions 1 and 2 are directly connected.
• Total WAN-BW-limits: enter “:NoLimit” to indicate that there is no explicit limit on
the bandwidth or number of simultaneous calls between the regions.
change ip-network-region 1
Page
3 of
19
Inter Network Region Connection Management
src
rgn
1
1
1
1
dst codec direct
Total
rgn set
WAN WAN-BW-limits
1
1
2
2
y
:NoLimit
3
4
Video
Norm Prio
Shr Intervening-regions
n
Dyn
CAC IGAR
n
Figure 12: IP Network Region 1 – Page 3
Configure IP Network Region 2, using the “change ip-network-region 2” command.
Enter:
• Name: a descriptive string such as “AcmePacket SIP Trks”
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
10 of 41
cm4AcmeSipPstn
•
•
•
•
Authoritative Domain: enter an IP address (or domain name) used to reach this network
region. In this case the IP address of the Acme Packet SIP interface (e.g.,
“150.100.100.130”) is used.
Codec Set: the value “2” corresponding to the ip-codec-set defined in Section 3.1.3 for
calls using the SIP PSTN gateway.
Intra-region IP-IP Direct Audio: the value “yes” (the default).
Inter-region IP-IP Direct Audio: the value “yes” (the default).
change ip-network-region 2
Page
1 of
19
IP NETWORK REGION
Region: 2
Location:
Authoritative Domain: 150.100.100.130
Name: AcmePacket SIP Trks
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 2
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 20000
IP Audio Hairpinning? n
UDP Port Max: 20999
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
Figure 13: IP Network Region 2 – Page 1
Verify that Page 3 of the “change ip-network-region 2” command appears as shown below
without any additional entries. The codec set and inter-region connectivity characteristics for the
src rgn 2 dst rgn 1 row were established during the configuration of IP network region 1.
change ip-network-region 2
Page
3 of
19
Inter Network Region Connection Management
src
rgn
2
2
2
2
dst codec direct
Total
rgn set
WAN WAN-BW-limits
1
2
y
:NoLimit
2
2
3
4
Video
Norm Prio
Shr Intervening-regions
n
Dyn
CAC IGAR
n
Figure 14: IP Network Region 2 – Page 3
3.1.7. Define SIP Trunk Groups
Two SIP trunk groups are defined for calls with the SIP PSTN gateway (routed via the Acme
Packet SD). Each SIP trunk group requires a corresponding SIP signaling group to define the
characteristics of the signaling relationship with the Acme Packet SD. Each signaling group uses
a separate C-LAN card for redundancy and capacity purposes.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
11 of 41
cm4AcmeSipPstn
All incoming calls use the round robin load balancing feature of the Acme Packet SD to
uniformly distribute calls across both C-LANs (and thus both SIP trunk groups). If the Acme
Packet SD detects a failure of SIP signaling to one C-LAN, it automatically routes all calls to the
remaining C-LAN interface until the failed signaling is restored.
All outbound calls are routed to the SIP trunk groups using Automatic Route Selection. The
route patterns selected by ARS overflow from the first choice SIP trunk group to the second
when a signaling failure or all trunks busy occurs.
3.1.7.1 Obtain “init” login ID for Avaya Communication Manager
The SIP signaling groups in these Application Notes require “tcp” as the transport method,
instead of the default “tls”. The use of “tcp” is administratively restricted and not an available
choice when using the “craft” or customer administrative login ids.
In order to complete the administration below, the “init” login id must be used to initially create
the signaling group. “Init” login privileges must be obtained from Avaya technical support.
After the signaling group is created the additional “init” privileges are no longer required (as
long as the transport method is not modified.
3.1.7.2 Establish the SIP Signaling Groups
Log into the Avaya Communcation Manager SAT using the “init” login ID.
Using the “add signaling-group n” command (where “n” is the number of the signaling group),
configure the signaling groups 3 and 4 as follows:
• Group Type: set to “sip”.
• Transport Method: set to “tcp”. (As discussed above, the “init” login id privileges are
required to perform this step. The Transport Method field will not be editable
otherwise.)
• Near-end Node Name: set to the C-LAN node name (defined in Section 3.1.2) used for
the respective signaling group. In these Application Notes, “clan_01a03” and
“clan_01a04” are used for signaling group 3 and 4, respectively.
• Far-end Node Name: set to the interface on the Acme Packet SD that will receive the
SIP signaling messages. In these Application Notes, “acme_cm4-side” will be used for
both signaling group 3 and 4. The IP address associated with this Far-end Node Name
will be the destination IP address where SIP messages are sent.
• Near-end Listen Port: set to “5060”, the default port of SIP signaling using tcp
transport.
• Far-end Listen Port: set to “5060”.
• Far-end Network Region: set to “2”, the network region defined for SIP PSTN gateway
calls defined in Section 3.1.6.
• Far-end Domain: set to IP address or domain name of the Acme Packet SD interface
used by Avaya Communication Manager. In these Application Notes the IP address of
“150.100.100.130” is used.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
12 of 41
cm4AcmeSipPstn
•
•
Direct IP-IP Audio Connections: set to “y”, indicating the RTP paths should be
optimized to reduce the use of media processing resources when possible.
DTMF over IP: set to “rtp-payload”. This value enables Avaya Communication
Manager to send DTMF transmissions using RFC 2833 [9].
The default values for the other fields may be used.
The resulting form for signaling group 3 is shown below.
add signaling-group 3
Page
1 of
1
SIGNALING GROUP
Group Number: 3
Group Type: sip
Transport Method: tcp
Near-end Node Name: clan_01a03
Near-end Listen Port: 5060
Far-end Node Name: acme_cm4-side
Far-end Listen Port: 5060
Far-end Network Region: 2
Far-end Domain: 150.100.100.130
Bypass If IP Threshold Exceeded? n
DTMF over IP: rtp-payload
Direct IP-IP Audio Connections? y
IP Audio Hairpinning? n
Enable Layer 3 Test? n
Session Establishment Timer(min): 3
Figure 15: Signaling Group 3
The resulting form for signaling group 4 is shown below.
add signaling-group 4
Page
1 of
1
SIGNALING GROUP
Group Number: 4
Group Type: sip
Transport Method: tcp
Near-end Node Name: clan_01a04
Near-end Listen Port: 5060
Far-end Node Name: acme_cm4-side
Far-end Listen Port: 5060
Far-end Network Region: 2
Far-end Domain: 150.100.100.130
Bypass If IP Threshold Exceeded? n
DTMF over IP: rtp-payload
Direct IP-IP Audio Connections? y
IP Audio Hairpinning? n
Enable Layer 3 Test? n
Session Establishment Timer(min): 3
Figure 16: Signaling Group 4
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
13 of 41
cm4AcmeSipPstn
3.1.7.3 Establish SIP Trunk Groups
Using the “add trunk-group n” command (where “n” is the number of the trunk group), configure
trunk groups 3 and 4.
On Page 1 of the Trunk Group form:
• Group Type: set to “sip”.
• Group Name: enter a descriptive string such as “SIP-AcmeTG3” and “SIP-AcmeTG4”
for trunk groups 3 and 4, respectively.
• TAC: enter a trunk access code such as “#003” and “#004” for trunk groups 3 and 4,
respectively.
• Service Type: set to “public-ntwrk” for trunks to the PSTN.
• Signaling Group: set to “3” and “4” (for trunk groups 3 and 4, respectively) as defined
within Section 3.1.7.2.
• Number of Members: set to the maximum number of simultaneous calls permitted for
each trunk group. Within these Application Notes, “10” was used for each trunk group.
The default values may be used on the remaining pages of the trunk-group form.
The resulting form for trunk-group 3 is shown below.
add trunk-group 3
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
3
SIP-AcmeTG3
two-way
n
0
public-ntwrk
Group Type: sip
COR: 1
Outgoing Display? n
TN: 1
CDR Reports: y
TAC: #003
Night Service:
Auth Code? n
Signaling Group: 3
Number of Members: 10
Figure 17: Trunk Group 3
The resulting form for trunk-group 4 is shown below.
add trunk-group 4
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
4
SIP-AcmeTG4
two-way
n
0
public-ntwrk
Group Type: sip
COR: 1
Outgoing Display? n
TN: 1
CDR Reports: y
TAC: #004
Night Service:
Auth Code? n
Signaling Group: 4
Number of Members: 10
Figure 18: Trunk Group 4
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
14 of 41
cm4AcmeSipPstn
3.1.8. Configure Calling Party Number Information
The calling party number (e.g., “18002224001”) is sent in the userinfo portion of the SIP “From”
header as shown below.
From: "Jane Smith" <sip:18002224001@150.100.100.130>;tag=80f839da25c3db
The “public-unknown-numbering” command controls the calling party number sent in the SIP
“From” field for calls originating from Avaya Communication Manager. The public-unknownnumbering is configured to send an 11 digit number consisting of “1800222” plus the 4 digit
extension number. In these Application Notes, extensions use numbers between “4000” and
“6999”.
Using the “change public-unknown-numbering n” command (where “n” is the leading digit of
the extension range), specify the calling party number information as follows:
•
•
•
•
•
Ext Len: set to “4”, the length of the extensions used.
Ext Code: set to the leading digit of the extension used. In these note, “4”, “5”, and “6”
are entered to cover all possible extensions between 4000 and 6999.
Trk Grp(s): by default, leave blank to perform the same conversion across all SIP (and
ISDN) trunk groups.
CPN Prefix: set to the leading digits (e.g., “1800222”) that are to be sent as the calling
party number.
Total CPN Len: set to the total length (e.g., “11”) of the calling party number to be sent.
The extension number will be appended to the CPN Prefix to form complete calling
party number of Total CPN Len digits.
The completed public-unknown-numbering form is shown below.
change public-unknown-numbering 4
Page
1 of
2
NUMBERING - PUBLIC/UNKNOWN FORMAT
Total
Ext Ext
Trk
CPN
CPN
Len Code
Grp(s)
Prefix
Len
Total Administered: 3
4 4
1800222
11
Maximum Entries: 9999
4 5
1800222
11
4 6
1800222
11
Figure 19: Public Unknown Numbering
3.1.9. Configure Call Routing
3.1.9.1 Outbound Calls
In these Application Notes, Automatic Route Selection is used to route outbound calls via the
SIP trunk groups to the Acme Packet SD (that in turn routes the calls to the PTSN gateway). In
addition the ARS route patterns support alternate routing (via the second SIP trunk group) should
the primary trunk group be unavailable.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
15 of 41
cm4AcmeSipPstn
Here, the configuration of one outbound calling pattern supporting calls to 1-733-xxx-xxx is
shown. Routing will select SIP trunk group 3 as the first choice, with overflow to SIP trunk
group 4 as required.
A typical installation will generally require additional dial string and route pattern entries but that
is beyond the scope of these Application Notes. Further information on ARS administration is
discussed in References [1] and [3].
ARS administration begins by verifying the availability of the feature as shown in Section 3.1.1.
Following the verification, use the “change dialplan analysis” command to create a feature
access code (fac) for ARS use.
• Dialed String: enter “9” that will become the user dialed prefix for outbound calls.
• Total Length: enter “1” as the length of the prefix.
• Call Type: enter “fac” as the type of prefix.
change dialplan analysis
Page
1 of
12
DIAL PLAN ANALYSIS TABLE
Percent Full:
Dialed
String
0
4
5
6
9
*
#
Total
Length
1
4
4
4
1
3
4
Call
Type
attd
ext
ext
ext
fac
dac
dac
Dialed
String
Total Call
Length Type
Dialed
String
1
Total Call
Length Type
Figure 20: Dial Plan Analysis
Use the “change feature-access-codes” command to assign the feature access code “9” to Auto
Route Selection (ARS) - Access Code 1 as shown below.
change feature-access-codes
Page
1 of
7
FEATURE ACCESS CODE (FAC)
Abbreviated Dialing List1 Access Code:
Abbreviated Dialing List2 Access Code:
Abbreviated Dialing List3 Access Code:
Abbreviated Dial - Prgm Group List Access Code:
Announcement Access Code: *71
Answer Back Access Code:
Auto Alternate Routing (AAR) Access Code:
Auto Route Selection (ARS) - Access Code 1: 9
Automatic Callback Activation:
Call Forwarding Activation Busy/DA: *61
All: *62
Call Forwarding Enhanced Status:
Act:
Call Park Access Code:
Call Pickup Access Code:
CAS Remote Hold/Answer Hold-Unhold Access Code:
CDR Account Code Access Code:
Change COR Access Code:
Change Coverage Access Code:
Contact Closure
Open Code:
Access Code 2:
Deactivation:
Deactivation: *60
Deactivation:
Close Code:
Figure 21: ARS Feature Access Code
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
16 of 41
cm4AcmeSipPstn
Use the “change ars analysis nn” command to configure the ARS route pattern selection rules as
follows. Here “nn” is “17”, the first two digits of the dialed number after the ARS access code.
• Dialed String: enter the leading digits (e.g., “1733”) necessary to uniquely select the
desired route pattern.
• Total Min: enter the minimum number of digits (e.g., “11”) expected for this PSTN
number.
• Total Max: enter the maximum number of digits (e.g., “11”) expected for this PSTN
number.
• Route Pattern: enter the route pattern number (e.g., “3”) to be used. The route pattern
(to be defined next) will specify the trunk group(s) to be used calls matching the dialed
number.
• Call Type: enter “fnpa”, the call type for North American 1+10 digit calls.
change ars analysis 17
Page
ARS DIGIT ANALYSIS TABLE
Location: all
Dialed
String
1733
Total
Min Max
11
11
Route
Pattern
3
Call
Type
fnpa
1 of
Percent Full:
Node
Num
2
1
ANI
Reqd
n
Figure 22: ARS Digit Analysis Entries
Use the “change route-pattern n” command (where “n” is the Route Pattern number used
above) to specify the SIP trunk groups selected for the outbound call.
In the form:
• Pattern Name: enter a descriptive string such as “SIP-AcmeSD-3,4” to describe the
routing pattern.
• Secure SIP?: leave as “n”, the default.
• Grp No: enter the trunk groups to be used in priority order. Trunk group 3 is the first
choice route followed by trunk group 4 in this configuration.
• FRL: enter the minimum facility restriction level (e.g., 1) necessary to use this trunk
group, with 0 being the least restrictive. The FRL within the Class of Restriction (COR)
assigned to the station must be greater than or equal to 1 in this case to use these trunk
groups.
• Pfx Mrk: enter “1”, to always send the prefix 1 on 10 digit calls.
• LAR: enter the routing behavior to be followed if the call is not successfully completed
using the trunk group. “Next” will cause the call to attempt to use the next choice in the
routing pattern. “None” indicates that no further attempts will be made to complete the
call. In the example below, a call that fails when attempting to use trunk group 3, will
automatically attempt to use trunk group 4 before being abandoned.
The defaults values for the remaining fields may be used.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
17 of 41
cm4AcmeSipPstn
The completed route pattern form is shown below.
change route-pattern 3
Pattern Number: 3
Grp FRL NPA Pfx Hop Toll No.
No
Mrk Lmt List Del
Dgts
1: 3
1
1
2: 4
1
1
3:
4:
5:
6:
1:
2:
3:
4:
5:
6:
Page
Pattern Name: SIP-AcmeSD-3,4
Secure SIP? n
Inserted
Digits
BCC VALUE TSC CA-TSC
0 1 2 M 4 W
Request
ITC BCIE Service/Feature PARM
y
y
y
y
y
y
rest
rest
rest
rest
rest
rest
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
y
n
n
n
n
n
n
n
n
n
n
n
n
1 of
3
DCS/
QSIG
Intw
n
n
n
n
n
n
IXC
user
user
user
user
user
user
No. Numbering LAR
Dgts Format
Subaddress
next
none
none
none
none
none
Figure 23: Route Pattern 3
3.1.9.2 Incoming Calls
This step configures the routing of incoming DID numbers to the proper extensions.
In these Application Notes, the following incoming toll-free 800 numbers are used.
Digits Received
(within SIP INVITE message)
800 222 4000
800 222 4001
800 222 5000
800 222 41xx
800 223 0000
Extension (or Hunt Group) Answering
4000
4001
5000
60xx
Forwarded to PSTN @ 1 732 555 1234 via SIP
trunk
Use the “change inc-call-handling-trmt trunk-group n” command (where “n” is the SIP trunk
group number) to administer the incoming number routing. This administration must be done for
each incoming trunk group.
• Called Len: enter the total number of incoming digits received (e.g., “10”).
• Called Number: enter the specific digit pattern to be matched.
• Del: enter the number of leading digits that should be deleted
• Insert: enter the specific digits to be inserted at the beginning of the adjusted incoming
digit string (to form what should be complete number).
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
18 of 41
cm4AcmeSipPstn
The completed inc-call-handling-trmt form for trunk group 3 is shown below.
change inc-call-handling-trmt trunk-group 3
INCOMING CALL HANDLING TREATMENT
Service/
Called
Called
Del Insert
Feature
Len
Number
public-ntwrk
10 80022240
6
public-ntwrk
10 80022241
8
60
public-ntwrk
10 8002225000
10 5000
public-ntwrk
10 8002230000
10 917325551234
Page
1 of
30
Page
1 of
30
The form for trunk group 4 is shown below.
change inc-call-handling-trmt trunk-group 4
INCOMING CALL HANDLING TREATMENT
Service/
Called
Called
Del Insert
Feature
Len
Number
public-ntwrk
10 80022240
6
public-ntwrk
10 80022241
8
60
public-ntwrk
10 8002230000
10 917325551234
3.1.10.
Save Avaya Communication Manager Changes
This completes the configuration of the Avaya Communication Manager.
Use the “save translation” command to make the changes permanent.
4. Configure the Acme Packet Net-Net Session Director
This section describes the configuration of the Acme Packet Net-Net Session Director. The NetNet Session Director acts as an intermediary between the Avaya Communication Manager
CLAN interfaces and the SIP PSTN gateway.
These Application Notes assume the Acme Packet SD has been previously installed according
Acme Packet guidelines. The basic installation and configuration is beyond the scope of this SIP
trunking application.
The Acme Packet Net-Net Session Director was configured using a telnet command line session
using its administrative interface. The following sections contain output (text outlined within
boxes) from the show running-config command that was used to display the configuration of
the SD. The general configuration information shown in Section 4.1 is not specific to the SIP
trunking in the Application Notes. It is included for reference purposes but without further
explanation. Section 4.2 through Section 4.4 contain the specific configuration details
(highlighted by bold text) important to the Direct SIP Trunking configuration within these
Application Notes. The remaining fields are generally the default value used by the Acme
Packet SD. For additional details on the administration of the Acme Packet SD, refer to [5].
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
19 of 41
cm4AcmeSipPstn
4.1. General Configuration
The general configuration elements of the Acme Packet SD configuration used are shown below
without further elaboration.
system-config
hostname
description
location
mib-system-contact
mib-system-name
mib-system-location
snmp-enabled
enable-snmp-auth-traps
enable-snmp-syslog-notify
enable-snmp-monitor-traps
enable-env-monitor-traps
snmp-syslog-his-table-length
snmp-syslog-level
system-log-level
process-log-level
process-log-ip-address
process-log-port
call-trace
internal-trace
log-filter
default-gateway
restart
exceptions
telnet-timeout
console-timeout
remote-control
last-modified-date
acmesystem
enabled
disabled
disabled
disabled
disabled
1
WARNING
WARNING
NOTICE
0.0.0.0
0
disabled
disabled
all
100.3.3.1
enabled
600
600
enabled
2007-01-30 11:27:07
host-routes
dest-network
netmask
gateway
last-modified-date
101.0.0.0
255.0.0.0
200.2.2.2
2006-08-17 15:35:15
redundancy-config
state
log-level
health-threshold
emergency-threshold
port
advertisement-time
percent-drift
initial-time
enabled
INFO
75
50
9090
500
210
1250
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
20 of 41
cm4AcmeSipPstn
becoming-standby-time
180000
becoming-active-time
100
cfg-port
1987
cfg-max-trans
10000
cfg-sync-start-time
5000
cfg-sync-comp-time
1000
gateway-heartbeat-interval
0
gateway-heartbeat-retry
0
gateway-heartbeat-timeout
1
gateway-heartbeat-health
0
peer
name
sbcsec
state
enabled
type
Secondary
destination
address
169.254.1.2:9090
network-interface
wancom1:0
destination
address
169.254.2.2:9090
network-interface
wancom2:0
peer
name
sbcpri
state
enabled
type
Primary
destination
address
169.254.1.1:9090
network-interface
wancom1:0
destination
address
169.254.2.1:9090
network-interface
wancom2:0
last-modified-date
2007-01-31 12:33:01
sip-config
state
operation-mode
dialog-transparency
home-realm-id
egress-realm-id
nat-mode
registrar-domain
registrar-host
registrar-port
init-timer
max-timer
trans-expire
invite-expire
inactive-dynamic-conn
pac-method
pac-interval
pac-strategy
pac-load-weight
pac-session-weight
JSR; Reviewed:
SPOC 4/20/2007
enabled
dialog
enabled
acme
Public
0
500
4000
32
180
32
10
PropDist
1
1
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
21 of 41
cm4AcmeSipPstn
pac-route-weight
pac-callid-lifetime
pac-user-lifetime
red-sip-port
red-max-trans
red-sync-start-time
red-sync-comp-time
add-reason-header
sip-message-len
last-modified-date
media-manager
state
latching
flow-time-limit
initial-guard-timer
subsq-guard-timer
tcp-flow-time-limit
tcp-initial-guard-timer
tcp-subsq-guard-timer
tcp-number-of-ports-per-flow
hnt-rtcp
algd-log-level
mbcd-log-level
home-realm-id
options
red-flow-port
red-mgcp-port
red-max-trans
red-sync-start-time
red-sync-comp-time
max-signaling-bandwidth
max-untrusted-signaling
min-untrusted-signaling
app-signaling-bandwidth
tolerance-window
rtcp-rate-limit
min-media-allocation
min-trusted-allocation
deny-allocation
last-modified-date
1
600
3600
1988
10000
5000
1000
disabled
0
2007-01-29 17:11:48
enabled
enabled
86400
300
300
86400
300
300
2
disabled
NOTICE
NOTICE
active-arp
1985
1986
10000
5000
1000
10000000
100
30
0
30
0
32000
1000
1000
2007-01-31 14:49:17
4.2. SIP Trunks to Avaya Communication Manager
4.2.1. Physical and Network Interfaces
In this configuration, a dedicated LAN subnet (150.100.100.0/24) was used for all
communication between the Acme Packet SD and the Avaya Communication Manager CLAN
and Medpro IP interfaces. This subnet does not have IP routing connectivity with any other IP
subnet. All SIP signaling and RTP media packets using this subnet are routed via Acme Packet
SD to reach the SIP PSTN gateway.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
22 of 41
cm4AcmeSipPstn
Ethernet interface slot 0 / port 2 (on each Acme Packet SD server in the redundant configuration)
is dedicated for the Avaya Communication Manager connectivity and connected to the Ethernet
switch providing the 150.100.100.0/24 LAN subnet.
The tables below show the details of the “phy-interface” and “network-interface” commands.
The key physical interface fields are:
• name: a descriptive string used to reference the Ethernet interface.
• operation-type: “Media” indicating both signaling and rtp packets use this interface.
• slot / port: the identifier of the specific front panel Ethernet interface used.
• virtual-mac: the mac address that will be dynamically assigned to the interface on the
active SD server (in the redundant pair). This address must be determined using the
guidelines provided in the Acme Packet SD documentation [6].
phy-interface
name
operation-type
port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
last-modified-date
cm4_clan
Media
2
0
00:08:25:01:b5:6e
enabled
enabled
FULL
100
2007-01-30 11:50:06
The key network-interface fields are:
• name: a reference of the phy-interface (defined above).
• ip-address: the virtual IP address of the active “cm4_clan” interface
• pri-utility-addr: the fixed IP address assigned to this interface on the primary SD server
• sec-utility-addr: the fixed IP address assigned to this interface on the secondary SD
server
• netmask: subnet mask for the IP subnet
• gateway: the subnet gateway address. In this case, gateway is null since all routing via
this interface remained within the 150.100.100.0/24 subnet.
• hip-ip-list: allowed ip address to accept administrative traffic (such as icmp ping)
• icmp-address: ip address used to pass icmp pings
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
23 of 41
cm4AcmeSipPstn
network-interface
name
sub-port-id
hostname
ip-address
pri-utility-addr
sec-utility-addr
netmask
gateway
sec-gateway
gw-heartbeat
state
heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
last-modified-date
cm4_clan
0
150.100.100.130
150.100.100.131
150.100.100.132
255.255.255.0
disabled
0
0
1
0
11
150.100.100.130
150.100.100.130
2007-01-31 18:36:18
4.2.2. SIP Interface
The “sip-interface” configuration defines the receiving characteristics of the SIP interfaces on the
Acme Packet SD. The Avaya Communication Manager SIP signaling groups will send SIP
messages to the sip-interface defined below.
The key sip-interface fields are:
• realm-id: the name (defined within Section 4.2.4) of the realm that this interface is
assigned.
• address: the ip address assigned to this sip-interface. This must match the IP Address
associated with the Name “acme_cm4_side” as defined in the IP Node Names form
shown in Figure 4. (Note this corresponds to the Far-end Node Name used in the SIP
signaling groups defined in Section 3.1.7.2.)
• port: the tcp port assigned to this sip-interface. This must match the Far-end Listen
Port assigned for the SIP signaling groups in Section 3.1.7.2.
• transport-protocol: the transport method used for this interface. This must match the
“tcp” Transport Method assigned for the signaling groups in Section 3.1.7.2.
sip-interface
state
realm-id
sip-port
address
JSR; Reviewed:
SPOC 4/20/2007
enabled
cm4_clan
150.100.100.130
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
24 of 41
cm4AcmeSipPstn
port
transport-protocol
tls-profile
allow-anonymous
carriers
proxy-mode
redirect-action
contact-mode
nat-traversal
nat-interval
tcp-nat-interval
registration-caching
min-reg-expire
registration-interval
route-to-registrar
secured-network
teluri-scheme
uri-fqdn-domain
trust-mode
max-nat-interval
nat-int-increment
nat-test-increment
sip-dynamic-hnt
stop-recurse
port-map-start
port-map-end
in-manipulationid
out-manipulationid
sip-ims-feature
operator-identifier
anonymous-priority
max-incoming-conns
per-src-ip-max-incoming-conns
inactive-conn-timeout
network-id
ext-policy-server
default-location-string
charging-vector-mode
charging-function-address-mode
ccf-address
ecf-address
term-tgrp-mode
implicit-service-route
rfc2833-payload
rfc2833-mode
last-modified-date
JSR; Reviewed:
SPOC 4/20/2007
5060
TCP
all
none
none
30
90
disabled
300
3600
disabled
disabled
disabled
all
3600
10
30
disabled
401,407
0
0
disabled
none
0
0
0
pass
pass
none
disabled
101
transparent
2007-01-30 11:21:53
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
25 of 41
cm4AcmeSipPstn
Steering pools define the range of UDP ports to be used for the RTP voice stream.
The key “steering-pool” parameters are:
• ip-address: the address of the interface on the Acme SD.
• start-port: an even number of the port that begins the range.
• end-port: an odd number of the port that ends the range.
• realm-id: the realm that that the steering pool is assigned to.
steering-pool
ip-address
start-port
end-port
realm-id
network-interface
last-modified-date
150.100.100.130
20000
20999
cm4_clan
2007-01-29 15:42:17
4.2.3. SIP Session Agent and Session Groups
The “session-agent” configuration defines specific interfaces where the Acme Packet SD will
send SIP signaling messages.
In this case, a “session-agent” is defined for each Avaya Communication Manager SIP signaling
group defined within Section 3.1.7.2. The field names below refer to fields in the signaling
group forms shown in Figure 15 and Figure 16 unless stated otherwise. Recall that these
signaling groups were defined use different C-LANs for reliability and load balancing purposes.
The key session-agent fields are:
• hostname: the IP address of the respective Near-end Node Name (e.g., respective CLANs). (This IP address is found on the IP Node Names form shown in Figure 4 in
Section 3.1.2.)
• port: the Near-end Listen Port specified for the respective signaling groups.
• app-protocol: enter “SIP”
• transport-method: “DynamicTCP” corresponding to the “TCP” Transport Method
specified in the signaling group forms.
• realm-id: the realm (e.g., “cm4_clan”) that these session agents belong to.
• description: a descriptive string to identify the far end interface
• max-sessions: a call admission control value that matches the Number of Members
value assigned in the trunk groups form shown in Figure 17 and Figure 18 in Section
3.1.7.3.
• ping-method: the specific SIP message sent to verify the sip trunk group is active. Note
this is a case-sensitive and must be entered exactly as “OPTIONS;hops=0”.
• ping-interval: the number of seconds between issuing the OPTIONS ping.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
26 of 41
cm4AcmeSipPstn
This is the session-agent configuration associated with the signaling group 3 (using C-LAN
1a03.)
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-outbound-sessions
max-burst-rate
max-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
last-modified-date
150.100.100.113
5060
enabled
SIP
DynamicTCP
cm4_clan
clan_1a03
enabled
enabled
10
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
OPTIONS;hops=0
60
disabled
disabled
0
disabled
none
0
2007-01-31 15:47:22
This is the session-agent configuration associated with the signaling group 4 (using C-LAN
1a04.)
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
27 of 41
cm4AcmeSipPstn
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-outbound-sessions
max-burst-rate
max-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
last-modified-date
150.100.100.114
5060
enabled
SIP
DynamicTCP
cm4_clan
clan_01a04
enabled
enabled
10
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
OPTIONS;hops=0
60
disabled
disabled
0
disabled
none
0
2007-01-31 15:47:30
The session-group configuration defines a logical group of session agents used to send SIP
messages to Avaya Communication Manager. Included in the session group configuration is the
strategy used to distribute calls across the multiple session agents. In this case a round-robin
strategy was chosen.
The key session-group fields are:
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
28 of 41
cm4AcmeSipPstn
•
•
•
•
•
group-name: a string used to reference this session group
description: a description of the session group
app-protocol: the signaling protocol used
strategy: the load balancing strategy used
dest: the hostname of the multiple session agents forming the session group
session-group
group-name
description
state
app-protocol
strategy
dest
cm4_sipTrkClans
all sip trunk clans
enabled
SIP
RoundRobin
150.100.100.113
150.100.100.114
trunk-group
last-modified-date
2007-01-26 17:29:06
4.2.4. Realm Configuration
The realm-config assigns common logical characteristics to be used by one or more interfaces,
address spaces, etc.
In this section the “cm4_clan” realm is defined. The primary function of this realm definition is
to uniformly apply rules to modify specific addresses within all outgoing SIP messages from the
interface associated with the realm. The sip message modification rules are defined within the
sip-manipulation configuration (named “NAT_IP”) shown in Section 4.4
The key realm-config fields in this configuration are:
• identifier: a string used as realm reference
• addr-prefix: the IP address subnet (e.g., “150.100.100.0/24”) that this realm applies to.
• network-interfaces: the name and sub-port-id (separated by a colon) of the network
interface(s) defined to be within this realm. These fields were specified in Section 4.2.1.
• out-manipulationid: enter the name of the “sip-manipulation” rule (defined in Section
4.4) that should be applied on messages sent by the realm.
The realm-config for “cm4_clan” (the subnet used for Avaya Communication Manager) is
shown below.
realm-config
identifier
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
msm-release
qos-enable
max-bandwidth
JSR; Reviewed:
SPOC 4/20/2007
cm4_clan
150.100.100.0/24
cm4_clan:0
disabled
enabled
enabled
enabled
disabled
disabled
0
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
29 of 41
cm4AcmeSipPstn
ext-policy-svr
max-latency
max-jitter
max-packet-loss
observ-window-size
parent-realm
dns-realm
media-policy
in-translationid
out-translationid
in-manipulationid
out-manipulationid
class-profile
average-rate-limit
access-control-trust-level
invalid-signal-threshold
maximum-signal-threshold
untrusted-signal-threshold
deny-period
symmetric-latching
pai-strip
trunk-context
early-media-allow
additional-prefixes
restricted-latching
restriction-mask
accounting-enable
last-modified-date
0
0
0
0
NAT_IP
0
none
0
0
0
30
disabled
disabled
none
32
enabled
2007-01-29 16:08:11
4.2.5. Local Policy
Local policy controls the routing of SIP calls from one realm to another.
The key local-policy fields in this configuration are:
• From-address: a policy filter indicating that the originating IP addresses that this policy
applies to. An asterisk (“*”) indicates any IP address.
• To-address: a policy filter indicating that the terminating IP addresses that this policy
applies to. An asterisk (“*”) indicates any IP address.
• Source-realm: a policy filter indicating the matching realm in order for the policy rules
to be applied.
• policy-attribute next-hop: the IP address where the message should be sent when the
policy rules match.
• policy-attribute realm: the realm associated with the next-hop IP address.
In this case, the policy provides a simple routing rule indicating that messages originating from
the “cm4_clan” realm are to be sent to the realm “pstn_cm4” via IP address 147.16.57.2 (the SIP
PSTN gateway).
This local-policy indicates that the Acme Packet SD should route calls originating from the
Avaya Communication Manager (“cm4_clan” realm) to the SIP PSTN gateway (“pstn_cm4”
realm) at IP address 142.16.57.2.
local-policy
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
30 of 41
cm4AcmeSipPstn
from-address
*
to-address
*
source-realm
activate-time
deactivate-time
state
policy-priority
last-modified-date
policy-attribute
next-hop
realm
action
terminate-recursion
carrier
start-time
end-time
days-of-week
cost
app-protocol
state
media-profiles
cm4_clan
N/A
N/A
enabled
none
2007-01-29 15:46:50
142.16.57.2
pstn_cm4
none
disabled
0000
2400
U-S
0
enabled
4.3. SIP Trunks to SIP PSTN Gateway
The configuration necessary for the Acme Packet SD to communicate with the SIP PSTN
gateway using SIP is shown below. Note that load balancing (using session agent groups) and
SIP failure detection (using SIP OPTION pings) were not defined for this gateway. The
description of the key fields was previously discussed in Section 4.2.1.
4.3.1. Physical and Network Interfaces
Here, Acme Packet SD interfaces at slot 0 / port 0 on each server are assigned addresses from the
LAN subnet (142.16.58.0/24) to communication with the SIP PSTN gateway at IP address
142.16.57.2.
The table below show the details of the “phy-interface”.
phy-interface
name
operation-type
port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
last-modified-date
JSR; Reviewed:
SPOC 4/20/2007
pstn
Media
0
0
00:08:25:01:be:e8
enabled
enabled
FULL
100
2006-03-08 17:40:04
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
31 of 41
cm4AcmeSipPstn
The table below shows the details of the “network-interface”.
network-interface
name
sub-port-id
hostname
ip-address
pri-utility-addr
sec-utility-addr
netmask
gateway
sec-gateway
gw-heartbeat
state
heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
last-modified-date
pstn
0
142.16.58.2
142.16.58.3
142.16.58.4
255.255.255.0
142.16.58.1
disabled
0
0
1
0
11
142.16.58.2
142.16.58.2
2006-10-10 08:44:01
4.3.2. SIP Interface
The “sip-interface” configuration defines the receiving characteristics of the SIP interface on the
Acme Packet SD for messages received from the SIP PSTN gateway. The description of the key
fields was previously discussed in Section 4.2.2.
This SIP interface uses the “UDP” transport-protocol and listens on the port “5060”. These
characteristics are matched to interface used on the SIP PSTN gateway in these Application
Notes.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
32 of 41
cm4AcmeSipPstn
Below is the complete sip-interface configuration.
sip-interface
state
realm-id
sip-port
address
port
transport-protocol
tls-profile
allow-anonymous
carriers
proxy-mode
redirect-action
contact-mode
nat-traversal
nat-interval
tcp-nat-interval
registration-caching
min-reg-expire
registration-interval
route-to-registrar
secured-network
teluri-scheme
uri-fqdn-domain
trust-mode
max-nat-interval
nat-int-increment
nat-test-increment
sip-dynamic-hnt
stop-recurse
port-map-start
port-map-end
in-manipulationid
out-manipulationid
sip-ims-feature
operator-identifier
anonymous-priority
max-incoming-conns
per-src-ip-max-incoming-conns
inactive-conn-timeout
network-id
ext-policy-server
default-location-string
charging-vector-mode
charging-function-address-mode
ccf-address
ecf-address
term-tgrp-mode
implicit-service-route
rfc2833-payload
rfc2833-mode
last-modified-date
JSR; Reviewed:
SPOC 4/20/2007
enabled
pstn_cm4
142.16.58.2
5060
UDP
all
none
none
30
90
disabled
300
3600
disabled
disabled
disabled
all
3600
10
30
disabled
401,407
0
0
disabled
none
0
0
0
pass
pass
none
disabled
101
transparent
2007-01-26 12:20:45
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
33 of 41
cm4AcmeSipPstn
Steering pools define the range of UDP ports to be used for the RTP voice stream to the SIP
PSTN gateway.
steering-pool
ip-address
start-port
end-port
realm-id
network-interface
last-modified-date
142.16.58.2
20000
20999
pstn_cm4
2007-01-29 15:42:26
4.3.3. SIP Session Agent and Session Groups
The “session-agent” configuration defines where SIP signaling messages will be sent. The
description of the key fields was previously discussed in Section 4.2.3
In this case, the “session-agent” defines the information used to send SIP messages to the SIP
PSTN gateway. Max sessions and ping-methods were not used for this SIP PSTN gateway.
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-outbound-sessions
max-burst-rate
max-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
media-profiles
in-translationid
JSR; Reviewed:
SPOC 4/20/2007
142.16.57.2
5060
enabled
SIP
UDP
*
pstn_gateway
enabled
disabled
0
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
0
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
34 of 41
cm4AcmeSipPstn
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
last-modified-date
disabled
disabled
0
disabled
none
0
2007-01-29 15:47:12
A session-group is not defined for the SIP PSTN gateway since load balancing was not part of
this configuration.
4.3.4. Realm Configuration
The realm-config assigns common logical characteristics to be used by one or more interfaces,
address spaces, etc. The description of the key fields was previously discussed in Section 4.2.4.
Below, the “pstn_cm4” realm is defined for communications with the SIP PSTN gateway. The
primary function of this realm definition is to apply the sip message modification rules are
defined within the sip-manipulation configuration (named “NAT_IP”) shown in Section 4.4.
The addr-prefix equal “0.0.0.0” means that the realm-config rules will be applied to messages
from any address received on the network-interfaces “pstn:0”.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
35 of 41
cm4AcmeSipPstn
realm-config
identifier
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
msm-release
qos-enable
max-bandwidth
ext-policy-svr
max-latency
max-jitter
max-packet-loss
observ-window-size
parent-realm
dns-realm
media-policy
in-translationid
out-translationid
in-manipulationid
out-manipulationid
class-profile
average-rate-limit
access-control-trust-level
invalid-signal-threshold
maximum-signal-threshold
untrusted-signal-threshold
deny-period
symmetric-latching
pai-strip
trunk-context
early-media-allow
additional-prefixes
restricted-latching
restriction-mask
accounting-enable
last-modified-date
pstn_cm4
0.0.0.0
pstn:0
disabled
enabled
enabled
enabled
disabled
disabled
0
0
0
0
0
NAT_IP
0
none
0
0
0
30
disabled
disabled
none
32
enabled
2007-01-29 16:08:27
4.3.5. Local Policy
Local policy controls the routing of SIP calls from one realm to another. The description of the
key fields was previously discussed in Section 4.2.5.
The local-policy indicates that the Acme Packet SD should route calls originating from the
“pstn_cm4” realm (that the SIP PSTN gateway is assigned to) using the Session Agent Group
(SAG) “cm4_sipTrkClans” defined within Section 4.2.3. Note the use of the Session Agent
Group indicator “SAG:” in the next-hop field. This indicates that messages use the logical
session group to perform the round robin load balancing rather than routing directly to a specific
C-LAN address. (The “SAG:” tag was not clearly documented in the Acme Packet
documentation in [6].)
local-policy
from-address
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
36 of 41
cm4AcmeSipPstn
*
to-address
*
source-realm
activate-time
deactivate-time
state
policy-priority
last-modified-date
policy-attribute
next-hop
realm
action
terminate-recursion
carrier
start-time
end-time
days-of-week
cost
app-protocol
state
media-profiles
pstn_cm4
N/A
N/A
enabled
none
2007-01-29 15:52:32
SAG:cm4_sipTrkClans
cm4_clan
none
disabled
0000
2400
U-S
0
SIP
enabled
4.4. SIP Header Manipulation
Sip-manipulation uses Header Manipulation Rules (HMR) to modify SIP messages according to
specific predefined rules.
In this application, the purpose is to convert addresses within SIP messages from the
150.100.100.0/24 subnet (used by Avaya Communication Manager) to the 142.16.57.0/24 subnet
(used by the SIP PSTN gateway.) when routing between the respective realms. This avoids the
alternative approach using the SIP NAT functions that can negatively impact performance and
capacity of the Acme Packet SD.
Further information can be found in the Acme Packet Best Current Practice Document 5200014-00 [6] that describe the use of HMR to improve performance.
The key sip-manipulation fields in this configuration are:
• name: a string used to reference the sip-manipulation rules (as used in the realm-config
in Sections 4.2.4 and 4.3.4).
• header-rule name: a descriptive string used to describe the header rule.
• header-rule msg-type: the type of SIP message that the rule is applied to.
• header-rule element-rule name: the specific SIP header the rule is applied to.
• header-rule element-rule new-value: “$LOCAL_IP” and “$REMOTE_IP” are variable
substitutions to be performed. “$LOCAL_IP” is the IP address of Acme Packet SD
interface sending the message. “$REMOTE_IP” is the address that the Acme Packet SD
is sending the message to.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
37 of 41
cm4AcmeSipPstn
The sip-manipulation rules used in these Application Notes is shown below.
sip-manipulation
name
NAT_IP
header-rule
name
action
match-value
msg-type
methods
element-rule
name
type
action
match-val-type
match-value
new-value
header-rule
name
action
match-value
msg-type
methods
element-rule
name
type
action
match-val-type
match-value
new-value
header-rule
name
action
match-value
msg-type
methods
element-rule
name
type
action
match-val-type
match-value
new-value
From
manipulate
request
FROM
uri-host
replace
ip
$LOCAL_IP
To
manipulate
request
TO
uri-host
replace
ip
$REMOTE_IP
Remote-Party-ID
manipulate
request
RPID
uri-host
replace
ip
$LOCAL_IP
5. Verification Steps
This section provides steps that may be performed to verify the operation of the Direct SIP
trunking configuration described in the Application Notes.
The Avaya Communication Manager “list trace station”, “list trace tac”, “status station” and/or
“status trunk-group” commands are helpful diagnostic tools to verify correct operation and to
troubleshoot problems. Also using a SIP protocol analyzer such as WireShark (a.k.a, Ethereal)
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
38 of 41
cm4AcmeSipPstn
to monitor the SIP messaging at the various interfaces (C-LAN, Acme Packet and/or SIP PSTN
gateway) is often very helpful in troubleshooting issues.
•
Incoming Calls – Verify that calls placed from a PSTN telephone to the DID number
assigned are properly routed via the SIP trunk group(s) to the expected telephone, hunt
group, ACD split, etc. Verify the talk-path exists in both directions, that calls remain
stable for several minutes and disconnect properly.
•
Outbound Calls – Verify that calls placed to a PSTN telephone are properly routed via
the SIP trunk group(s) defined in the ARS route patterns. Verify that the talk-path exists
in both directions and that calls remain stable and disconnect properly.
•
Direct IP-IP Connections – This applies if IP telephones and Direct IP-IP are used. Verify
that stable calls are using Direct IP-IP talk paths using the “status station” or “status
trunk-group” commands. When Direct IP-IP is used, the Audio Connection field will
indicate “ip-direct” instead of “ip-tdm”.
•
Load Balancing of Incoming Calls – This applies if multiple SIP trunk groups (using
multiple C-LANs and Acme Packet Load Balancing) are used. Verify that incoming calls
are distributed across the trunk groups defined with the session-agent-group of the Acme
Packet SD.
•
Alternate Routing of Inbound Calls on C-LAN failure – This applies if multiple SIP trunk
groups (using multiple C-LANs) are used. Maintenance busy the C-LAN associated with
an incoming SIP trunk group and verify using the “list trace station” or “list trace trunk”
commands that inbound calls are routed to the active SIP trunk group (using a separate
C-LAN). Verify that the original trunk group is used once the C-LAN is returned to
service. Repeat for other incoming SIP trunk groups. Note: This may be service
affecting!
•
Alternate Routing of Outbound Calls on C-LAN failure – This applies if multiple SIP
trunk groups (using multiple C-LANs) are used. Maintenance busy the C-LAN
associated with the first-choice trunk group and verify using the “list trace station” or
“list trace trunk” commands that outbound calls overflow to the next SIP trunk group
(using a separate C-LAN). Verify that the original trunk group is used once the C-LAN
is returned to service. Repeat for other route-patterns using these trunk groups. Note:
This may be service affecting!
6. Support
For technical support on the Acme Packet Net-Net Session Director, visit www.acmepacketcom.
7. Conclusion
These Application Notes describe the configuration steps required to establish sip trunking
directly with Avaya Communication Manager to an Acme Packet Net-Net Session Director and a
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
39 of 41
cm4AcmeSipPstn
SIP PSTN gateway for the purpose of PSTN interconnection. This configuration was
successfully compliance tested with the demonstration of calls in both directions with the PSTN.
The ability to use incoming load balancing across multiple Avaya Communication Manger CLAN interfaces and endure a C-LAN interface isolation or failure was shown.
8. References
The Avaya product documentation is available at http://support.avaya.com.
[1] Administrator Guide for Avaya Communication Manager, February 2007, Issue 3,
Document Number 03-300509.
[2] Adding New Hardware for Avaya Media Servers and Gateways, February 2007, Issue 2,
Release 4.0, Document Number 03-300684
[3] Feature Description and Implementation for Avaya Communication Manager, Issue 5,
Document Number 555-245-205
[4] SIP Support in Avaya Communication Manager Running on the Avaya S8300, S8400,
S8500 series and S8700 series Media Server, March 2007, Issue 6.1, Document Number
555-245-206.
[5] 4600 Series IP Telephone Release 2.6 LAN Administrator Guide, August 2006, Issue 4,
Document Number 555-233-507
The following documentation is provided with the Acme Packet Net-Net Session Director or is
available from Acme Packet Technical Support. See www.acmepacket.com for further
information.
[6] Net-Net Session Director Configuration Guide, Acme Packet, Inc., Release Version 4.1,
Document Number 400-0061-41A, August 8, 2006.
[7] SIP Peering Configuration on the Net-Net Session Director - Software Versions 4.0.0 and
Newer, Acme Packet Best Current Practice, Document # 520-0014-00, March 16, 2006
Several Internet Engineering Task Force (IETF) standards track RFC documents were
referenced within these Application Notes. The RFC documents may be obtained at:
http://www.rfc-editor.org/rfcsearch.html.
[8] RFC 3261 - SIP (Session Initiation Protocol), June 2002, Proposed Standard
[9] RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, May
2000, Proposed Standard
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
40 of 41
cm4AcmeSipPstn
©2007 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.
JSR; Reviewed:
SPOC 4/20/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All Rights Reserved.
41 of 41
cm4AcmeSipPstn
Was this manual useful for you? yes no
Thank you for your participation!

* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project

Download PDF

advertising