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Avaya Solution & Interoperability Test Lab
Application Notes for Avaya Aura® Communication
Manager 6.3, Avaya Aura® Session Manager 6.3, and Avaya
Session Border Controller for Enterprise 6.2 with Verizon
Business IP Contact Center (IPCC) Services Suite – Issue 1.1
Abstract
These Application Notes describe a sample configuration of Avaya Aura® Communication
Manager 6.3, Avaya Aura® Session Manager 6.3, and Avaya Session Border Controller for
Enterprise 6.2 with Verizon Business IP Contact Center (IPCC) Services suite. The Verizon
Business IPCC Services suite includes the IP Toll Free VoIP Inbound and IP-IVR SIP trunk service offers. This service suite provides toll free inbound calling via standards-based SIP trunks as well as re-routing of inbound toll free calls to alternate destinations based upon SIP messages (i.e., REFER) generated by Communication Manager. The Communication Manager
Network Call Redirection (NCR) and SIP User-to-User Information (UUI) features can be utilized together to transmit UUI within SIP signaling messages to alternate destinations via the Verizon network. These Application Notes update previously published Application Notes with newer versions of Communication Manager and Session Manager, and present an example configuration for the Avaya Session Border Controller for Enterprise.
Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted in the Avaya Solution &
Interoperability Test Lab, utilizing a Verizon Business Private IP (PIP) circuit connection to the production Verizon Business IPCC Services.
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Table of Contents
Inbound IP Toll Free Call with no Network Call Redirection ..................................11
Inbound IP Toll Free Call with Post-Answer Network Call Redirection .................12
Inbound IP Toll Free Call with Unsuccessful Network Call Redirection ................13
Configure Avaya Aura® Communication Manager Release 6.3 .............................15
Processor Ethernet Configuration on HP Common Server .......................................19
Post-Answer Redirection With UUI to a SIP Destination ........................................32
ACD Configuration for Call Queued for Handling by Agent ...................................33
Incoming Call Handling Treatment for Incoming Calls ...........................................36
Saving Communication Manager Configuration Changes .......................................37
Configure Avaya Aura® Session Manager Release 6.3 ...........................................38
Configure Avaya Session Border Controller for Enterprise Release 6.2 ..................54
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Verizon Business IPCC Services Suite Configuration .............................................81
Avaya Aura® Communication Manager Verifications ............................................82
Example Incoming Call from PSTN via Verizon IPCC to Telephone .....................82
Example Incoming Call Referred via Call Vector to PSTN Destination ..................83
Avaya Aura® System Manager and Avaya Aura® Session Manager Verifications86
Avaya Session Border Controller for Enterprise Verification ..................................89
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1. Introduction
These Application Notes describe a sample configuration of Avaya Aura® Communication
Manager 6.3, Avaya Aura® Session Manager 6.3, and Avaya Session Border Controller for
Enterprise with Verizon Business IP Contact Center (IPCC) Services suite. The Verizon Business
IPCC Services suite includes the IP Toll Free VoIP Inbound and IP-IVR SIP trunk service offers.
This service suite provides toll free inbound calling via standards-based SIP trunks as well as rerouting of inbound toll free calls to alternate destinations based upon SIP messages (i.e., REFER) generated by Communication Manager. The Communication Manager Network Call Redirection
(NCR) and SIP User-to-User Information (UUI) features can be utilized together to transmit UUI within SIP signaling messages to alternate destinations via the Verizon network. These Application
Notes update previously published Application Notes [VZ-IPTF] and [VZ-IP-IVR] with a newer version of Session Manager, Communication Manager, and Avaya SBCE.
In the sample configuration, an Avaya Session Border Controller for Enterprise (ASBCE) is used as an edge device between the Avaya CPE and Verizon Business. The Avaya SBCE performs SIP header manipulation and provides topology hiding to convert the private Avaya CPE IP addressing to IP addressing or domains appropriate for the Verizon access method. Avaya Aura® Session
Manager is used as the Avaya SIP trunking “hub” connecting to Communication Manager, the
Avaya SBCE, and other applications.
The Verizon Business IPCC Services suite described in these Application Notes is designed for business customers. The suite provides inbound toll-free service via standards-based SIP trunks.
Using SIP Network Call Redirection (NCR), trunk-to-trunk connections of certain inbound calls at
Communication Manager can be avoided by requesting that the Verizon network transfer the inbound caller to an alternate destination. In addition, the Communication Manager SIP User-to-
User Information (UUI) feature can be utilized with the SIP NCR feature to transmit UUI within
SIP signaling messages to alternate destinations. This capability allows the service to transmit a limited amount of call-related data between call centers to enhance customer service and increase call center efficiency. Examples of UUI data might include a customer account number obtained during a database query or the best service routing data exchanged between sites using
Communication Manager.
Verizon Business IPCC Services suite is a portfolio of IP Contact Center (IPCC) interaction services that includes VoIP Inbound and IP Interactive Voice Response (IP IVR). Access to these features may use Internet Dedicated Access (IDA) or Private IP (PIP). PIP was used for the sample configuration described in these Application Notes. VoIP Inbound is the base service offering that offers core call routing and termination features. IP IVR is an enhanced service offering that includes features such as menu-routing, custom transfer, and additional media capabilities.
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For more information on the Verizon Business IP Contact Center service, visit http://www.verizonbusiness.com/Products/communications/contact-center/
2. General Test Approach and Test Results
The Avaya equipment depicted in Figure 1 was connected to the commercially available Verizon
Business IPCC Services. This allowed PSTN users to dial toll-free numbers assigned by Verizon.
The toll-free numbers were configured to be routed within the enterprise to Avaya Aura®
Communication Manager numbers, including Vector Directory Numbers (VDNs). The VDNs were associated with vectors configured to exercise Communication Manager ACD functions as well as
Verizon IPCC Services such as network call redirection to PSTN destinations and network call redirection with UUI.
The test approach was manual testing of inbound and referred calls using the Verizon IPCC
Services on a production Verizon PIP access circuit, as shown in Figure 1.
The main objectives were to verify the following features and functionality:
Inbound Verizon toll-free calls to Communication Manager telephones and VDNs/Vectors
Inbound private toll-free calls (e.g., PSTN caller uses *67 followed by the toll-free number)
Inbound Verizon toll-free calls redirected using Communication Manager SIP NCR (via
SIP REFER/Refer-To) to PSTN alternate destinations
Inbound Verizon IP toll-free calls redirected using Communication Manager SIP NCR with
UUI (via SIP REFER/Refer-To with UUI) to a SIP-connected destination
Inbound toll-free voice calls can use G.711MU or G.729A codecs
Inbound toll-free voice calls can use DTMF transmission using RFC 2833
Testing was successful. Test observations or limitations are described in Section 2.2.
See Section 3.2 for an overview of key call flows and Section 9 for detailed verifications and traces illustrating key call flows.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or completeness of a DevConnect member’s solution.
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2.1. Interoperability Compliance Testing
The interoperability compliance testing included the execution of test cases details in the Verizonauthored interoperability test plan.
SIP OPTIONS monitoring of the health of the SIP trunks was verified. Both the Avaya enterprise equipment and Verizon Business can monitor health using SIP OPTIONS.
Incoming calls from the PSTN were routed to the toll-free numbers assigned by Verizon
Business to the Avaya location. Configuration was varied such that these incoming toll-free calls were directed to Communication Manager telephone extensions, and Communication
Manager VDNs containing call routing logic to exercise SIP Network Call Redirection.
Proper disconnect when either party hangs up an active call.
Proper disconnect when the PSTN caller abandons (i.e., hangs up) a toll free call before the call has been answered.
Proper SIP 486 response and busy tone heard by the caller when a PSTN user calls a tollfree number directed to a busy user or resource when no redirection on busy conditions was configured (which would be unusual in a contact center).
Proper termination of an inbound IP Toll Free call left in a ringing state for a relatively long duration, which again would be unusual in a contact center. In the sample configuration, Verizon sent a SIP CANCEL to cancel the call after three minutes of ring no answer conditions, returning busy tone to the PSTN caller.
Privacy requests for inbound toll-free calls from the PSTN were verified. That is, when privacy is requested by a PSTN caller (e.g., dialing *67 from a mobile phone), the inbound toll-free call can be successfully completed while withholding presentation of the PSTN caller id to user displays. (When the caller requests privacy, Verizon IPCC sends the caller
ID in the P-Asserted-Identity header and includes “Privacy: id” which is honored by
Communication Manager).
Inbound toll-free call long holding time call stability. The Avaya CPE sends a re-INVITE with SDP to refresh the session at the configured session refresh interval specified on the
Communication Manager trunk group handling the call. In the sample configuration, the session refresh re-INVITE was sent after 900 seconds (15 minutes), the interval configured for the trunk group in Section 5.8. The call continued with proper talk path.
Telephony features such as hold and resume. When a Communication Manager user holds a call in the sample configuration, Communication Manager will send a re-INVITE to
Verizon IPCC with a media attribute “sendonly”. The Verizon 200 OK to this re-INVITE will include media attribute “recvonly”. While the call remains on hold, RTP will flow from the Avaya CPE to Verizon, but no RTP will flow from Verizon to the Avaya CPE
(i.e., as intended). When the user resumes the call from hold, bi-directional media path resumes. Although it would be unexpected in a contact center, calls on hold for longer than the session refresh interval were tested, and such calls could be resumed after the session refresh re-asserted the “sendonly” state.
Transfer of toll-free calls between Communication Manager users.
Incoming voice calls using the G.729a and G.711 ULAW codecs, and proper protocol procedures related to media.
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DTMF transmission using RFC2833. For inbound toll-free calls, PSTN users dialing postanswer DTMF digits are recognized properly by the Avaya CPE.
Proper DiffServ markings for SIP signaling and RTP media flowing from the Avaya CPE to Verizon.
2.2. Test Results
The interoperability compliance testing of the sample configuration was completed with successful results as described in Section 2.1. The following observations may be noteworthy:
Verizon Business IPCC Services suite does not support fax.
Verizon Business IPCC Services suite does not support History Info or Diversion Headers. The
Avaya CPE will not send History-Info or Diversion header to Verizon IPCC in the sample configuration.
Verizon Business IPCC Services suite does not support G.729 Annex b. When using G729, the
Avaya CPE will always include “annexb=no” in SDP in the sample configuration.
Section 3.3.3 summarizes a call flow that would theoretically allow a call to remain in
Communication Manager vector processing upon failure of a vector-triggered REFER attempt.
However, such call scenarios could not be verified on the production Verizon circuit used for testing. On the production circuit, Verizon would send a BYE to terminate the call upon encountering REFER transfer failures, so there was no opportunity for the call to remain in
Communication Manager vector processing. See Section 3.3.3 for additional information.
The presence of unnecessary headers such as P-Location in a SIP message to Verizon does not cause any user-perceivable problems. Nevertheless, SBC procedures are shown in Section 7.7 to illustrate how headers such as P-Location that are not required by Verizon may be removed by the SBC.
2.3. Support
2.3.1 Avaya
For technical support on the Avaya products described in these Application Notes visit http://support.avaya.com
2.3.2 Verizon
For technical support on Verizon Business IPCC service offer, visit online support at http://www.verizonbusiness.com/us/customer/
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3. Reference Configuration
Figure 1 illustrates the sample configuration used for the DevConnect compliance testing. The configuration is comprised of the Avaya CPE location connected via a T1 Internet connection to the Verizon Business IPCC service node. The Avaya CPE location simulates a customer site. At the edge of the Avaya CPE location is an Avaya Session Border Controller for Enterprise. The
Avaya SBCE receives traffic from the Verizon Business IPCC Services on port 5060 and sends traffic to the Verizon Business IPCC Services using destination port 5072, using UDP for transport. The PIP service defines a secure MPLS connection between the Avaya CPE T1 connection and the Verizon IPCC service node.
Figure 1: Avaya Interoperability Test Lab Configuration
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The Verizon toll-free numbers were mapped by Session Manager or Communication Manager to various Communication Manager extensions. The extension mappings were varied during the testing to allow inbound toll-free calls to terminate directly on user extensions or indirectly through hunt groups, vector directory numbers (VDNs) and vectors to user extensions and contact center agents.
The Avaya CPE environment was known to Verizon Business IPCC Service as FQDN
adevc.avaya.globalipcom.com. For efficiency, the Avaya CPE environment utilizing Session
Manager Release 6.3 and Communication Manager Release 6.3 was shared among other ongoing test efforts at the Avaya Solutions and Interoperability Test lab. Access to the Verizon Business
IPCC services was added to a configuration that already used domain “avayalab.com” at the enterprise. As such, the Avaya SBCE is used to adapt the domains as needed. These Application
Notes indicate the configuration that would not be required in cases where the CPE domain in
Communication Manager and Session Manager match the CPE domain known to Verizon.
The following summarizes various header contents and manipulations for toll-free calls in the sample configuration:
Verizon Business IPCC Services node sends the following in the initial INVITE to the
CPE: o The CPE FQDN of adevc.avaya.globalipcom.com in the Request URI. o The Verizon IPCC Services gateway IP address in the From header. o The enterprise SBC outside IP address (i.e., 1.1.1.2) in the To header. o Sends the INVITE to Avaya CPE using destination port 5060 via UDP
Avaya Session Border Controller for Enterprise sends Session Manager: o The Request URI contains avayalab.com. o The host portion of the From header and PAI header contains avayalab.com o The host portion of the To header contains avayalab.com o Sends the packet to Session Manager using destination port 5060 via TCP
Session Manager sends Communication Manager o The Request URI contains avayalab.com, to match the shared Avaya SIL test environment. o Sends the packet to Communication Manager using destination port 5071 via TLS to allow Communication Manager to distinguish Verizon traffic from other traffic arriving from the same instance of Session Manager.
Note – The Fully Qualified Domain Names and IP addressing specified in these Application Notes apply only to the reference configuration shown in Figure 1. Verizon Business customers will use
FQDNs and IP addressing appropriate for the unique customer environment.
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3.1. Remote Workers
In the sample configuration, remote Avaya SIP endpoints connected through Avaya SBCE with
Advanced Services licensing were used along with local Avaya endpoints in the verification of these Application Notes. The figure below illustrates a detailed view of the Remote Workers section previously shown in Figure 1. Although not the primary focus of these Application Notes, relevant configuration parameters of the Avaya SBCE for use with Remote Worker are illustrated in Appendix A.
Figure 2: Remote Worker Lab Configuration
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3.2. History Info and Diversion Headers
The Verizon Business IPCC Services suite does not support SIP History Info Headers or Diversion
Headers. Therefore, Communication Manager was provisioned not to send History Info Headers or
Diversion Headers.
3.3. Call Flows
To understand how inbound Verizon toll-free calls are handled by Session Manager and
Communication Manager, key call flows are summarized in this section.
3.3.1 Inbound IP Toll Free Call with no Network Call Redirection
to Communication Manager, which in turn routes the call to a vector, agent, or phone. No redirection is performed in this simple scenario. A detailed verification of such a call with
Communication Manager traces can be found in Section 9.1.1.
1. A PSTN phone originates a call to a Verizon IP Toll Free number.
2. The PSTN routes the call to the Verizon IP Toll Free service network.
3. The Verizon IP Toll Free service routes the call to the Avaya Session Border Controller for
Enterprise.
4. The Avaya Session Border Controller for Enterprise performs any configured SIP header modifications, and routes the call to Session Manager.
5. Session Manager applies any configured SIP header adaptations and digit conversions, and
6. based on configured Routing Policies, determines where the call should be routed. In this case, Session Manager routes the call to Communication Manager using a unique port so that Communication Manager can distinguish this call as having arrived from Verizon
IPCC.
Depending on the called number, Communication Manager routes the call to a) a hunt group or vector, which in turn routes the call to an agent or phone, or b) directly to a phone.
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1
PSTN
2
Verizon
IP Toll Free Service
PSTN
Phone
Avaya Aura®
Communication
Manager
6
5
3
Avaya Session
Border
Controller
4
Avaya Aura®
Session
Manager
MESSAGE
OK
PHONE
CONTACTS MENU CALL LOG
DEF .,@ ABC
GHI JKL MNO
FORWARD
HEADSET
SPEAKER
PQRS
*
TUV WXYZ
VOLUME
MUTE
Agent / Phone
Figure 3: Inbound Verizon IP Toll Free Call – No Redirection
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3.3.2 Inbound IP Toll Free Call with Post-Answer Network Call Redirection
The second call scenario illustrated in Figure 4 is an inbound Verizon IP Toll Free call that is
routed to a Communication Manager Vector Directory Number (VDN) to invoke call handling logic in a vector. The vector answers the call and then redirects the call back to the Verizon IP Toll
Free service for routing to an alternate destination. Note that Verizon IP Toll Free service does not support redirecting a call before it is answered (using a SIP 302), and therefore the vector must include a step that results in answering the call, such as playing an announcement, prior to redirecting the call using REFER.
A detailed verification of such call with Communication Manager traces can be found in Section
9.1.2 for a Verizon IP Toll Free SIP-connected alternate destination. In this example, the Verizon
IP Toll Free service can be used to pass User to User Information (UUI) from the redirecting site to the alternate destination.
1. Same as the first five steps in Figure 3.
2. Communication Manager routes the call to a vector, which answers the call, plays an announcement, and attempts to redirect the call by sending a SIP REFER message out the
SIP trunk from which the inbound call arrived. The SIP REFER message specifies the alternate destination in the Refer-To header. The SIP REFER message passes back through
Session Manager and the Avaya SBCE to the Verizon IP Toll Free service network.
3. The Verizon IP Toll Free service places a call to the target party contained in the Refer-To header. Upon answer, the calling party is connected to the target party.
4. The Verizon IP Toll Free service notifies the Avaya CPE that the referred call has been answered (NOTIFY/sipfrag 200 OK). Communication Manager sends a BYE. The calling party and the target party can talk. The trunk upon which the call arrived in Step 1 is idle.
1
PSTN
1
Verizon
IP Toll Free
Service
3
Alternate
Destination
3
OK
PHONE MESSAGE
CONTACTS MENU
ABC .,@
GHI JKL
CALL LOG
DEF
MNO
FORWARD
HEADSET
PQRS
*
SPEAKER
TUV WXYZ
VOLUME
MUTE
Agent / Phone
PSTN
Phone
1 2 4
Avaya Session
Border
Controller
1 2 4
1
Avaya Aura®
Communication
Manager
2
Avaya Aura®
Session
Manager
4
Figure 4: Inbound Verizon IP Toll Free– Post-Answer SIP REFER Redirection Successful
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3.3.3 Inbound IP Toll Free Call with Unsuccessful Network Call Redirection
The next call scenario illustrated in Figure 5 is similar to the previous call scenario, except that the redirection is unsuccessful. In theory, if redirection is successful, Communication Manager can
“take the call back” and continue vector processing. For example, the call may route to an alternative agent, phone, or announcement after unsuccessful NCR.
1. Same as Figure 4.
2. Same as Figure 4.
3. The Verizon IP Toll Free service places a call to the target party (alternate destination), but the target party is busy or otherwise unavailable.
4. The Verizon IP Toll Free service notifies the redirecting/referring party (Communication
Manager) of the error condition.
5. Communication Manager routes the call to a local agent, phone, or announcement.
However, as noted in Section 2.2, except for egregious configuration errors, this “REFER error handling” scenario could not be verified on the production Verizon circuit used for testing. On the production circuit, Verizon sends a SIP BYE which terminates Communication Manager vector processing for failure scenarios. For example, if a 486 Busy is received from the target of the
REFER, Verizon will send a BYE immediately after a “NOTIFY/sipfrag 486”, which precludes any further call processing by Communication Manager. As another example, in cases where misconfiguration is introduced to cause the Refer-To header to be malformed (e.g., no “+” in Refer-
To), Verizon will send a BYE immediately after a “NOTIFY/sipfrag 603 Server Internal Error”. If
REFER is configured in the vector, but Network Call Redirection is not enabled for the SIP trunk group, Communication Manager will not send the REFER to Verizon, and vector processing will continue at the step following the route-to step that would normally trigger the REFER.
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1
PSTN
1
Verizon
IP Toll Free
Service
3
Alternate
Destination
3
MESSAGE
OK
PHONE
FORWARD
PQRS
HEADSET
SPEAKER
*
CONTACTS MENU CALL LOG
ABC DEF .,@
GHI JKL MNO
TUV WXYZ
VOLUME
MUTE
Agent / Phone
PSTN
Phone
Avaya Aura®
Communication
Manager
1
2
4
5
1 2 4
Avaya Session
Border
Controller
1 2
Avaya Aura®
Session
Manager
4
MESSAGE
OK
PHONE
CONTACTS MENU
.,@ ABC
CALL LOG
DEF
GHI JKL MNO
TUV WXYZ FORWARD PQRS
HEADSET
SPEAKER
*
VOLUME
MUTE
Agent / Phone
Figure 5: Inbound Verizon IP Toll Free– Post-Answer SIP REFER Redirection Unsuccessful
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4. Equipment and Software Validated
The following equipment and software were used in the sample configuration.
Equipment:
HP ProLiant DL360 G7
Software:
Avaya Aura® Communication
Manager Release 6.3 SP0
HP ProLiant DL360 G7
Avaya Aura® System Manager 6.3
SP2
HP ProLiant DL360 G7
Avaya Aura® Session Manager 6.3
SP2
G450 Gateway 33.13.0
DELL 210 RII
Avaya 9600-Series Telephones (H.323)
Avaya 96X1- Series Telephones (SIP)
Avaya 96X1- Series Telephones (H323)
Avaya One-X Communicator (H.323)
Avaya Session Border Controller for
Enterprise Version 6.2 Q36
R 3.2
R6.2.2.17
R6.2313
6.1.8.06-SP8-40314
Avaya Flare® Experience for Windows
Avaya Desktop Video Device
Avaya 6400-Series Digital Telephones
1.1.2.11
Flare 1.1.3
N/A
Table 1: Equipment and Software Used in the Sample Configuration
5.
Configure Avaya Aura® Communication Manager Release 6.3
This section illustrates an example configuration allowing SIP signaling via the “Processor
Ethernet” of Communication Manager to Session Manager. In configurations that use an Avaya
G650 Media Gateway, it is also possible to use an Avaya C-LAN in the Avaya G650 Media
Gateway for SIP signaling to Session Manager.
Note - The initial installation, configuration, and licensing of the Avaya servers and media gateways for Communication Manager are assumed to have been previously completed and are not discussed in these Application Notes. These Application Notes focus on describing the sample configuration as it relates to SIP Trunking to Verizon IPCC.
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5.1. Verify Licensed Features
Communication Manager license file controls customer capabilities. Contact an authorized Avaya representative for assistance if a required feature needs to be enabled.
On Page 2 of the display system-parameters customer-options form, verify that the Maximum
Administered SIP Trunks is sufficient for the combination of trunks to the Verizon Business
IPCC Services and any other SIP applications. Each call from the Verizon Business IPCC Services to a non-SIP endpoint uses one SIP trunk for the duration of the call. Each call from Verizon
Business IPCC Services to a SIP endpoint uses two SIP trunks for the duration of the call.
display system-parameters customer-options Page 2 of 11
OPTIONAL FEATURES
IP PORT CAPACITIES USED
Maximum Administered H.323 Trunks: 12000 0
Maximum Concurrently Registered IP Stations: 18000 3
Maximum Administered Remote Office Trunks: 12000 0
Maximum Concurrently Registered Remote Office Stations: 18000 0
Maximum Concurrently Registered IP eCons: 128 0
Max Concur Registered Unauthenticated H.323 Stations: 100 0
Maximum Video Capable Stations: 36000 3
Maximum Video Capable IP Softphones: 18000 1
Maximum Administered SIP Trunks: 12000 40
Maximum Administered Ad-hoc Video Conferencing Ports: 12000 0
Maximum Number of DS1 Boards with Echo Cancellation: 522 0
Maximum TN2501 VAL Boards: 10 0
Maximum Media Gateway VAL Sources: 250 2
Maximum TN2602 Boards with 80 VoIP Channels: 128 0
Maximum TN2602 Boards with 320 VoIP Channels: 128 0
Maximum Number of Expanded Meet-me Conference Ports: 300 0
On Page 4 of the display system-parameters customer-options form, verify that the IP Trunks and IP Stations features are enabled. If the use of SIP REFER messaging or send-only SDP attributes will be required verify that the ISDN/SIP Network Call Redirection feature is enabled.
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OPTIONAL FEATURES
Emergency Access to Attendant? y IP Stations? y
Enable 'dadmin' Login? y
Enhanced Conferencing? y ISDN Feature Plus? n
Enhanced EC500? y ISDN/SIP Network Call Redirection? y
Enterprise Survivable Server? n ISDN-BRI Trunks? y
Enterprise Wide Licensing? n ISDN-PRI? y
ESS Administration? y Local Survivable Processor? n
Extended Cvg/Fwd Admin? y Malicious Call Trace? y
External Device Alarm Admin? y Media Encryption Over IP? n
Five Port Networks Max Per MCC? n Mode Code for Centralized Voice Mail? n
Flexible Billing? n
Forced Entry of Account Codes? y Multifrequency Signaling? y
Global Call Classification? y Multimedia Call Handling (Basic)? y
Hospitality (Basic)? y Multimedia Call Handling (Enhanced)? y
Hospitality (G3V3 Enhancements)? y Multimedia IP SIP Trunking? y
IP Trunks? y
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On Page 5 of the display system-parameters customer-options form, verify that the Private
Networking and Processor Ethernet features are enabled.
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OPTIONAL FEATURES
Multinational Locations? n Station and Trunk MSP? y
Multiple Level Precedence & Preemption? n Station as Virtual Extension? y
Multiple Locations? n
System Management Data Transfer? n
Personal Station Access (PSA)? y Tenant Partitioning? y
PNC Duplication? n Terminal Trans. Init. (TTI)? y
Port Network Support? y Time of Day Routing? y
Posted Messages? y TN2501 VAL Maximum Capacity? y
Uniform Dialing Plan? y
Private Networking? y Usage Allocation Enhancements? y
Processor and System MSP? y
Processor Ethernet? y Wideband Switching? y
Wireless? n
Remote Office? y
Restrict Call Forward Off Net? y
Secondary Data Module? y
Subnet Mask: /24
On Page 6 of the System-Parameters Customer-Options form, verify that any required call center features are enabled. In the sample configuration, vectoring is used to refer calls to alternate destinations using SIP NCR. Vector variables are used to include User-User Information (UUI) with the referred calls.
display system-parameters customer-options Page 6 of 11
CALL CENTER OPTIONAL FEATURES
Call Center Release: 6.0
ACD? y Reason Codes? y
BCMS (Basic)? y Service Level Maximizer? n
BCMS/VuStats Service Level? y Service Observing (Basic)? y
BSR Local Treatment for IP & ISDN? y Service Observing (Remote/By FAC)? y
Business Advocate? n Service Observing (VDNs)? y
Call Work Codes? y Timed ACW? y
DTMF Feedback Signals For VRU? y Vectoring (Basic)? y
Dynamic Advocate? n Vectoring (Prompting)? y
Expert Agent Selection (EAS)? y Vectoring (G3V4 Enhanced)? y
EAS-PHD? y Vectoring (3.0 Enhanced)? y
Forced ACD Calls? n Vectoring (ANI/II-Digits Routing)? y
Least Occupied Agent? y Vectoring (G3V4 Advanced Routing)? y
Lookahead Interflow (LAI)? y Vectoring (CINFO)? y
Multiple Call Handling (On Request)? y Vectoring (Best Service Routing)? y
Multiple Call Handling (Forced)? y Vectoring (Holidays)? y
PASTE (Display PBX Data on Phone)? y Vectoring (Variables)? y
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On Page 7 of the System-Parameters Customer-Options form, verify that the required call center capacities can be met. In the sample configuration, agents will log in (using agent-login IDs) to staff the ACD and handle inbound calls from Verizon IP Toll Free.
display system-parameters customer-options Page 7 of 11
CALL CENTER OPTIONAL FEATURES
VDN of Origin Announcement? y VuStats? y
VDN Return Destination? y VuStats (G3V4 Enhanced)? y
USED
Logged-In ACD Agents: 5200 1
Logged-In Advocate Agents: 5200 0
Logged-In IP Softphone Agents: 5200 0
Logged-In SIP EAS Agents: 500 0
5.2. Dial Plan
In the reference configuration the Avaya CPE environment uses five digit local extensions, such as
12xxx, 14xxx or 20xxx. Trunk Access Codes (TAC) are 3 digits in length and begin with *. The
Feature Access Code (FAC) to access ARS is the single digit 9. The Feature Access Code (FAC) to access AAR is the single digit 8. The dial plan illustrated here is not intended to be prescriptive; any valid dial plan may be used.
The dial plan is modified with the change dialplan analysis command as shown below.
change dialplan analysis Page 1 of 12
DIAL PLAN ANALYSIS TABLE
Location: all Percent Full: 1
Dialed Total Call Dialed Total Call Dialed Total Call
String Length Type String Length Type String Length Type
1 5 ext
2 5 ext
8 1 fac
9 1 fac
* 3 dac
# 3 dac
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5.3. Node Names
Node names are mappings of names to IP addresses that can be used in various screens. The following change node-names ip output shows relevant node-names in the sample configuration.
As shown in bold, the node name for Session Manager is “SM63” with IP address 10.64.19.226.
The node name and IP address for the Processor Ethernet “procr” is 10.64.19.155.
change node-names ip Page 1 of 2
IP NODE NAMES
Name IP Address
SM63 10.64.19.226
default 0.0.0.0
procr 10.64.19.155
procr6 ::
5.4. Processor Ethernet Configuration on HP Common Server
The add ip-interface procr or change ip-interface procr command can be used to configure the
Processor Ethernet (PE) parameters. The following screen shows the parameters used in the sample configuration. While the focus here is the use of the PE for SIP Trunk Signaling, observe that the Processor Ethernet will also be used for registrations from H.323 IP Telephones and H.248 gateways in the sample configuration.
change ip-interface procr Page 1 of 2
IP INTERFACES
Type: PROCR
Target socket load: 1700
Enable Interface? y Allow H.323 Endpoints? y
Allow H.248 Gateways? y
Network Region: 1 Gatekeeper Priority: 5
IPV4 PARAMETERS
Node Name: procr IP Address: 10.64.19.155
Subnet Mask: /24
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5.5. Network Regions for Gateway, Telephones
Network regions provide a means to logically group resources. In the shared Communication
Manager configuration used for the testing, the Avaya G450 Media Gateway is in region 1. To provide testing flexibility, network region 2 was associated with other components used specifically for the Verizon testing.
Non-IP telephones (e.g., analog, digital) derive network region and location configuration from the
Avaya gateway to which the device is connected. The following display command shows that
Media Gateway 1 is an Avaya G450 Media Gateway configured for network region 1. It can also be observed that the Controller IP Address is the Avaya Processor Ethernet (10.64.19.155), and that the gateway IP address is 10.64.19.81. These fields are not configured in this screen, but just display the current information for the Media Gateway.
change media-gateway 1 Page 1 of 2
MEDIA GATEWAY 1
Type: g450
Name: G450-1
Serial No: 08IS38199678
Encrypt Link? y Enable CF? n
Network Region: 1 Location: 1
Site Data:
Recovery Rule: 1
Registered? y
FW Version/HW Vintage: 33 .13 .0 /1
MGP IPV4 Address: 10.64.19.81
MGP IPV6 Address:
Controller IP Address: 10.64.19.155
MAC Address: 00:1b:4f:03:52:18
Subnet Mask: /24
The following screen shows Page 2 for Media Gateway 1. The gateway has an S8300 in slot V1
(unused), an MM712 media module supporting Avaya digital phones in slot V2, an MM711 supporting analog devices in slot V3, and the capability to provide announcements and music on hold via “gateway-announcements” in logical slot V9.
change media-gateway 1 Page 2 of 2
MEDIA GATEWAY 1
Type: g450
Slot Module Type Name DSP Type FW/HW version
V1: S8300 ICC MM MP80 110 3
V2: MM712 DCP MM
V3: MM711 ANA MM
V4:
V5:
V6:
V7:
V8: Max Survivable IP Ext: 8
V9: gateway-announcements ANN VMM
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IP telephones can be assigned a network region based on an IP address mapping. The network region can also associate the IP telephone to a location for location-based routing decisions. The following screen illustrates a subset of the IP network map configuration used to verify these
Application Notes. If the IP address of a registering IP Telephone does not appear in the ipnetwork-map, the phone is assigned the network region of the “gatekeeper” (e.g., CLAN or PE) to which it registers. When the IP address of a registering IP telephone is in the ip-network-map, the phone is assigned the network region assigned by the form shown below. For example, the IP address 10.64.19.109 would be mapped to network region 1, based on the configuration in bold below. In production environments, different sites will typically be on different networks, and ranges of IP addresses assigned by the DHCP scope serving the site can be entered as one entry in the network map, to assign all telephones in a range to a specific network region.
change ip-network-map Page 1 of 63
IP ADDRESS MAPPING
Subnet Network Emergency
IP Address Bits Region VLAN Location Ext
--------------------------------------------- ------ ------ ---- -------------
FROM: 10.64.19.100 / 1 n
TO: 10.64.19.120
The following screen shows IP Network Region 2 configuration. In the shared test environment, network region 2 is used to allow unique behaviors for the Verizon IPCC test environment. In this example, codec set 2 will be used for calls within region 2. The shared Avaya Interoperability Lab test environment uses the domain “avayalab.com” (i.e., for network region 1 including the region of the Processor Ethernet “procr”). Session Manager also uses this domain to determined routes for calls based on the domain information of the calls and for SIP phone registration.
change ip-network-region 2 Page 1 of 20
IP NETWORK REGION
Region: 2
Location: 1 Authoritative Domain: avayalab.com
Name: Session Manager Stub Network Region: n
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 2 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 46
Audio PHB Value: 46
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
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The following screen shows the inter-network region connection configuration for region 2. The first bold row shows that network region 2 is directly connected to network region 1, and that codec set 2 will also be used for any connections between region 2 and region 1. For configurations where multiple remote gateways are used, each gateway will typically be configured for a different region, and this screen can be used to specify unique codec or call admission control parameters for the pairs of regions. If a different codec should be used for interregion connectivity than for intra-region connectivity, a different codec set can be entered in the
codec set column for the appropriate row in the screen shown below. Once submitted, the configuration becomes symmetric, meaning that network region 1, Page 4 will also show codec set
2 for region 2 to region 1 connectivity.
change ip-network-region 2 Page 4 of 20
Source Region: 2 Inter Network Region Connection Management I M
G A t
dst codec direct WAN-BW-limits Video Intervening Dyn A G c
rgn set WAN Units Total Norm Prio Shr Regions CAC R L e
1 2 y NoLimit n t
2 2 all
3
4
The following screen shows IP Network Region 1 configuration. In this example, codec set 1 will be used for calls within region 1 due to the Codec Set parameter on Page 1, but codec set 2 will be used for connections between region 1 and region 2 as noted previously.
change ip-network-region 1 Page 1 of 20
IP NETWORK REGION
Region: 1
Location: 1 Authoritative Domain: avayalab.com
Name: Enterprise Stub Network Region: n
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
Call Control PHB Value: 46
Audio PHB Value: 46
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
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The following screen shows the inter-network region connection configuration for region 1. The bold row shows that network region 1 is directly connected to network region 2, and that codec set
2 will be used for any connections between region 2 and region 1.
change ip-network-region 1 Page 4 of 20
Source Region: 1 Inter Network Region Connection Management I M
G A t
dst codec direct WAN-BW-limits Video Intervening Dyn A G c
rgn set WAN Units Total Norm Prio Shr Regions CAC R L e
1 1 all
2 2 y NoLimit n t
5.6. IP Codec Sets
The following screen shows the configuration for codec set 2, the codec set configured to be used for calls within region 2 and for calls between region 1 and region 2. In general, an IP codec set is a list of allowable codecs in priority order. Using the example configuration shown below, all calls with Verizon IPCC via the SIP trunks would prefer to use G.729A, but also be capable of using
G.711MU (The Verizon IPCC service will not include G.722 in SDP offers or SDP answers). Any calls using this same codec set that are between devices capable of the G.722-64K codec can use
G.722. The specification of G.722 as the first choice is not required. That is, G.722 may be omitted from the codec set, but it is recommended that G.729A and G.711MU be included in the codec set for use with Verizon IPCC Services.
change ip-codec-set 2 Page 1 of 2
IP Codec Set
Codec Set: 2
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.722-64K 2 20
2: G.729A n 2 20
3: G.711MU n 2 20
4:
On Page 2 of the form, configure the FAX Mode field to off. Verizon IPCC does not support fax.
change ip-codec-set 2 Page 2 of 2
IP Codec Set
Allow Direct-IP Multimedia? n
Mode Redundancy
FAX off 0
Modem off 0
TDD/TTY US 3
Clear-channel n 0
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Although codec set 1 is not used for connections with Verizon IPCC, the following screen shows the configuration for codec set 1. Codec set 1 is used for local Avaya CPE connections within region 1. change ip-codec-set 1 Page 1 of 2
IP Codec Set
Codec Set: 1
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.722.2 n 1 20
2: G.722-64K 2 20
3: G.711MU n 2 20
4:
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5.7. SIP Signaling Group
This section illustrates the configuration of the SIP Signaling Groups. Each signaling group has a
Group Type of “sip”, a Near-end Node Name of “procr”, and a Far-end Node Name of
“SM63”. In the example screens, the Transport Method for all signaling groups is “tls”. The Peer
Detection Enabled field is set to “y” and a peer Session Manager has been previously detected.
The Far-end Domain is set to “avayalab.com” matching the configuration in place prior to adding the Verizon IP SIP Trunking configuration. The Enable Layer 3 Test field is enabled on each of the signaling groups to allow Communication Manager to maintain the signaling group using the
SIP OPTIONS method. Fields that are not referenced in the text below can be left at default values, including DTMF over IP set to “rtp-payload”, which corresponds to RFC 2833.
The following screen shows signaling group 2. Signaling group 2 will be used for processing incoming calls from Verizon IPCC via Session Manager. The Far-end Network Region is configured to region 2. Port 5071 has been configured as both the Near-end Listen Port and Far-
end Listen Port. Session Manager will be configured to direct calls arriving from the PSTN with
Verizon toll-free numbers to a route policy that uses a SIP entity link to Communication Manager specifying port 5071. The use of different ports is one means to allow Communication Manager to distinguish different types of calls arriving from the same Session Manager. Other parameters may be left at default values.
change signaling-group 2 Page 1 of 2
SIGNALING GROUP
Group Number: 2 Group Type: sip
IMS Enabled? n Transport Method: tls
Q-SIP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM
Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? y
Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? n
Near-end Node Name: procr Far-end Node Name: SM63
Near-end Listen Port: 5071 Far-end Listen Port: 5071
Far-end Network Region: 2
Far-end Domain: avayalab.com
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6
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The following screen shows signaling group 3, the signaling group to Session Manager that was in place prior to adding the Verizon IPCC configuration to the shared Avaya Solutions and
Interoperability Test Lab configuration. This signaling group reflects configuration not specifically related to Verizon IPCC but will be used to enable SIP phones to register to Session Manager and to use features from Communication Manager. Again, the Near-end Node Name is “procr” and the Far-end Node Name is “SM63”, the node name of the Session Manager. Unlike the signaling group used for the Verizon IPCC signaling, the Far-end Network Region is “1”. The Peer
Detection Enabled field is set to “y” and a peer Session Manager has been previously detected.
change signaling-group 3 Page 1 of 2
SIGNALING GROUP
Group Number: 3 Group Type: sip
IMS Enabled? n Transport Method: tls
Q-SIP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM
Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? y
Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? n
Near-end Node Name: procr Far-end Node Name: SM63
Near-end Listen Port: 5061 Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain: avayalab.com
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6
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5.8. SIP Trunk Group
This section illustrates the configuration of the SIP Trunk Groups corresponding to the SIP signaling group from the previous section.
NOTE: For Verizon Business customers utilizing either Verizon IP Contact Center or IP-IVR service offers, at least one Elite Agent license is required to support the ability to utilize the
Network Call Redirection capabilities of those services with Communication Manager. This license is required to enable the ISDN/SIP Network Call Redirection feature. This licensed feature must be turned ON to support Network Call Redirection. Additional details on how to configure Network Call Redirection in Communication Manager can be found within the supporting text and figures contained within this section.
The following shows Page 1 for trunk group 2, which will be used for incoming and outgoing
PSTN calls from Verizon. The Number of Members field defines how many simultaneous calls are permitted for the trunk group. The Service Type field is set to “public-ntwrk” for the trunks that will handle calls with Verizon. Although not strictly necessary, the Direction has been configured to “incoming” to emphasize that trunk group 1 is used for incoming calls only in the sample configuration.
change trunk-group 2 Page 1 of 21
TRUNK GROUP
Group Number: 2 Group Type: sip CDR Reports: y
Group Name: VerizonIPCC COR: 1 TN: 1 TAC: *02
Direction: incoming Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: public-ntwrk Auth Code? n
Member Assignment Method: auto
Signaling Group: 2
Number of Members: 10
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The following screen shows Page 2 for trunk group 2. All parameters shown are default values, except for the Preferred Minimum Session Refresh Interval, which has been changed from the default 600 to 900. Although not strictly necessary, some SIP products prefer a higher session refresh interval than Communication Manager default value, which can result in unnecessary SIP messages to re-establish a higher refresh interval for each call.
change trunk-group 2 Page 2 of 21
Group Type: sip
TRUNK PARAMETERS
Unicode Name: auto
Redirect On OPTIM Failure: 5000
SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 900
Disconnect Supervision - In? y Out? y
XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n
The following screen shows Page 3 for trunk group 2. All parameters except those in bold are default values. The Numbering Format will use “private” numbering, meaning that the private numbering table would be consulted for any mappings of Communication Manager extensions to alternate numbers to be sent to Session Manager. Replacement text strings can be configured using the “system-parameters features” screen (page 9, not shown), such that incoming “private”
(anonymous) or “restricted” calls can display a configurable text string on called party telephones.
If it is desired to see the configurable replacement text strings on user displays, the Replace
Restricted Numbers and Replace Unavailable Numbers fields may be set to “y”.
change trunk-group 2 Page 3 of 21
TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y
Numbering Format: private
UUI Treatment: service-provider
Replace Restricted Numbers? y
Replace Unavailable Numbers? y
Modify Tandem Calling Number: no
Show ANSWERED BY on Display? y
The following screen shows Page 4 for trunk group 2. The bold fields have non-default values.
The Convert 180 to 183 for Early Media field was a new in Communication Manager Release 6.
Verizon recommends that inbound calls to the enterprise result in a 183 with SDP rather than a 180 with SDP, and setting this field to “y” for the trunk group handling inbound calls from Verizon produces this result. Although not strictly necessary, the Telephone Event Payload Type has been set to 101 to match Verizon configuration. Setting the Network Call Redirection flag to “y”
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enables advanced services associated with the use of the REFER message, while also implicitly enabling Communication Manager to signal “send-only” media conditions for calls placed on hold at the enterprise site. If neither REFER signaling nor “send-only” media signaling is required, this field may be left at the default “n” value. In the testing associated with these Application Notes, the Network Call Redirection flag was set to “y” to allow REFER to be exercised with the
Verizon IPCC Service.
The Verizon IPCC Services do not support the Diversion header or the History-Info header, and therefore both Support Request History and Send Diversion Header are set to “n”
change trunk-group 2 Page 4 of 21
PROTOCOL VARIATIONS
Mark Users as Phone? n
Prepend '+' to Calling/Alerting/Diverting/Connected Number? n
Send Transferring Party Information? n
Network Call Redirection? y
Build Refer-To URI of REFER From Contact For NCR? n
Send Diversion Header? n
Support Request History? n
Telephone Event Payload Type: 101
Convert 180 to 183 for Early Media? y
Always Use re-INVITE for Display Updates? n
Identity for Calling Party Display: P-Asserted-Identity
Block Sending Calling Party Location in INVITE? n
Accept Redirect to Blank User Destination? n
Enable Q-SIP? n
The following screen shows Page 1 for trunk group 3, the bi-directional “tie” trunk group to
Session Manager that existed before adding the Verizon SIP Trunk configuration to the shared
Avaya Interoperability Lab network. Recall that this trunk is used to enable SIP phones to use features from Communication Manager and to communicate with other Avaya applications, such as Avaya Aura® Messaging, and does not reflect any unique Verizon configuration.
change trunk-group 3 Page 1 of 21
TRUNK GROUP
Group Number: 3 Group Type: sip CDR Reports: y
Group Name: To SM Enterprise COR: 1 TN: 1 TAC: *03
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: auto
Signaling Group: 3
Number of Members: 20
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The following shows Page 3 for trunk group 3. Note that this tie trunk group uses a “private”
Numbering Format.
change trunk-group 3 Page 3 of 21
TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y
Numbering Format: private
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Modify Tandem Calling Number: no
The following screen shows Page 4 for trunk group 3. Note that unlike the trunks associated with
Verizon calls that have non-default “protocol variations”, this trunk group maintains all default values. Support Request History must remain set to the default “y” to support proper subscriber mailbox identification by Modular Messaging.
change trunk-group 3 Page 4 of 21
PROTOCOL VARIATIONS
Mark Users as Phone? n
Prepend '+' to Calling/Alerting/Diverting/Connected Number? n
Send Transferring Party Information? n
Network Call Redirection? n
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 120
Convert 180 to 183 for Early Media? y
Always Use re-INVITE for Display Updates? n
Identity for Calling Party Display: P-Asserted-Identity
Block Sending Calling Party Location in INVITE? n
Accept Redirect to Blank User Destination? n
Enable Q-SIP? n
5.9. Contact Center Configuration
This section describes the basic commands used to configure Vector Directory Numbers (VDNs) and corresponding vectors. These vectors contain steps that invoke the Communication Manager
SIP Network Call Redirection (NCR) functionality. These Application Notes provide rudimentary vector definitions to demonstrate and test the SIP NCR and UUI functionalities. In general, call centers will use vector functionality that is more complex and tailored to individual needs. Call centers may also use customer hosts running applications used in conjunction with Application
Enablement Services (AES) to define call routing and provide associated UUI. The definition and documentation of those complex applications and associated vectors are beyond the scope of these
Application Notes.
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5.9.1 Announcements
Various announcements will be used within the vectors. In the sample configuration, these announcements were sourced by the Avaya G450 Media Gateway. The following abridged list command summarizes the announcements used in conjunction with the vectors in this section. To add an announcement extension, use the command “add announcement <extension>”. list announcement
ANNOUNCEMENTS/AUDIO SOURCES
Announcement Source Num of
Extension Type Name Pt/Bd/Grp Files
11001 integrated callcenter-main 001V9 1
11002 integ-mus holdmusic 001V9 1
11003 integrated disconnect 001V9 1
11004 integrated no_agents 001V9 1
11005 integrated dtmf_test 001V9 1
11006 integrated please_wait 001V9 1
11007 integrated REFER_Test 001V9 1
5.9.2 Post-Answer Redirection to a PSTN Destination
This section provides an example configuration of a vector that will use post-answer redirection to a PSTN destination. A corresponding detailed verification is provided in Section 9.2.2. In this example, the inbound toll-free call is routed to VDN 10001 shown in the following screen. The originally dialed Verizon IP Toll Free number may be mapped to VDN 3698 by Session Manager digit conversion, or via the incoming call handling treatment for the Communication Manager trunk group handling the call.
display vdn 10001 Page 1 of 3
VECTOR DIRECTORY NUMBER
Extension: 10001
Name*: Refer-to-PSTN
Destination: Vector Number 1
Attendant Vectoring? n
Meet-me Conferencing? n
Allow VDN Override? n
COR: 1
TN*: 1
Measured: none
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VDN 10001 is associated with vector 1, which is shown below. Vector 1 plays an announcement
(step 03) to answer the call. After the announcement, the “route-to number” (step 05) includes
“~r+13035387024” where the number 303-538-7024 is a PSTN destination. This step causes a
REFER message to be sent where the Refer-To header includes “+13035387024” as the user portion. Note that Verizon IP Contact Center services require the “+” in the Refer-To header for this type of call redirection.
display vector 1 Page 1 of 6
CALL VECTOR
Number: 1 Name: Refer-to-PSTN
Multimedia? n Attendant Vectoring? n Meet-me Conf? n Lock? n
Basic? y EAS? y G3V4 Enhanced? y ANI/II-Digits? y ASAI Routing? y
Prompting? y LAI? y G3V4 Adv Route? y CINFO? y BSR? y Holidays? y
Variables? y 3.0 Enhanced? y
01 wait-time 2 secs hearing ringback
02 # Play announcement to caller in step 3. This answers the call.
03 announcement 11006
04 # Refer the call to PSTN Destination in step 5 below.
05 route-to number ~r+13035387024 with cov n if unconditionally
06 # If Refer fails queue to skill 1
07 queue-to skill 1 pri m
08
5.9.3 Post-Answer Redirection With UUI to a SIP Destination
This section provides an example of post-answer redirection with UUI passed to a SIP destination.
A corresponding detailed verification is provided in Section 9.2.3. In this example, the inbound call is routed to VDN 10003 shown in the following screen. The originally dialed Verizon toll-free number may be mapped to VDN 10003 by Session Manager digit conversion, or via the incoming call handling treatment for the Communication Manager trunk group handling the call.
display vdn 10003 Page 1 of 3
VECTOR DIRECTORY NUMBER
Extension: 10003
Name*: REFER with UUI
Destination: Vector Number 3
Attendant Vectoring? n
Meet-me Conferencing? n
Allow VDN Override? n
COR: 1
TN*: 1
Measured: none
To facilitate testing of NCR with UUI, the following vector variables were defined.
change variables Page 1 of 39
VARIABLES FOR VECTORS
Var Description Type Scope Length Start Assignment VAC
A uui asaiuui L 16 1
B uui asaiuui L 16 17
C
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VDN 10003 is associated with vector 3, which is shown below. Vector 3 sets data in the vector variables A and B (steps 03 and 04) and plays an announcement to answer the call (step 05). After the announcement, the “route-to” number step includes “~r+18668512649”. This step causes a
REFER message to be sent where the Refer-To header includes “+18668512649” as the user portion. The Refer-To header will also contain the UUI set in variables A and B. Verizon will include this UUI in the INVITE ultimately sent to the SIP-connected target of the REFER, which is toll-free number “18668512649”. In the sample configuration, where only one location was used, 866-851-2649 is another toll-free number assigned to the same circuit as the original call. In practice, NCR with UUI would allow Communication Manager to send call or customer-related data along with the call to another contact center.
display vector 3 Page 1 of 6
CALL VECTOR
Number: 3 Name: Refer-with-UUI
Multimedia? n Attendant Vectoring? n Meet-me Conf? n Lock? n
Basic? y EAS? y G3V4 Enhanced? y ANI/II-Digits? y ASAI Routing? y
Prompting? y LAI? y G3V4 Adv Route? y CINFO? y BSR? y Holidays? y
Variables? y 3.0 Enhanced? y
01 wait-time 2 secs hearing ringback
02 set A = none CATR 1234567890123456
03 set B = none CATR 7890123456789012
04 # Play announcement to answer call and route to ~r to cause Refer
05 announcement 11007
06 route-to number ~r+18668512649 with cov n if unconditionally
07 # If Refer fails play announcement and disconnect
08 disconnect after announcement 11003
5.9.4 ACD Configuration for Call Queued for Handling by Agent
This section provides a simple example configuration for VDN, vector, hunt group, and agent logins used to queue inbound Verizon IPCC calls for handling by an agent.
The following screens show an example ACD hunt group. On page 1, note the bolded values.
display hunt-group 1 Page 1 of 4
HUNT GROUP
Group Number: 1 ACD? y
Group Name: Agent Group Queue? y
Group Extension: 19991 Vector? y
Group Type: ucd-mia
TN: 1
COR: 1 MM Early Answer? n
Security Code: Local Agent Preference? n
ISDN/SIP Caller Display:
Queue Limit: unlimited
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The following screens show an example ACD hunt group. On the abbreviated page 2 shown below, note Skill is set to “y”.
display hunt-group 1 Page 2 of 4
HUNT GROUP
Skill? y Expected Call Handling Time (sec): 180
AAS? n Service Level Target (% in sec): 80 in 20
VDN 10004, shown below, is associated with vector 4.
display vdn 10004 Page 1 of 3
VECTOR DIRECTORY NUMBER
Extension: 10004
Name*: Sales
Destination: Vector Number 4
Attendant Vectoring? n
Meet-me Conferencing? n
Allow VDN Override? n
COR: 1
In this simple example, vector 4 briefly plays ring back, then queues the call to skill 1.
Announcement 11004 is a simple recurring announcement. If an agent is immediately available to handle the call, the call will be delivered to the agent. If an agent is not immediately available, the call will be queued, and the caller will hear the announcement. Once an agent becomes available, the call will be delivered to the agent.
display vector 4 Page 1 of 6
CALL VECTOR
Number: 4 Name: Sales
Multimedia? n Attendant Vectoring? n Meet-me Conf? n Lock? n
Basic? y EAS? y G3V4 Enhanced? y ANI/II-Digits? y ASAI Routing? y
Prompting? y LAI? y G3V4 Adv Route? y CINFO? y BSR? y Holidays? y
Variables? y 3.0 Enhanced? y
01 # Wait hearing ringback
02 wait-time 2 secs hearing ringback
03 # Simple queue to skill with recurring announcement until available
04 queue-to skill 1 pri m
05 announcement 11004
06 wait-time 30 secs hearing music
07 goto step 5 if unconditionally
08 stop
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The following screen illustrates an example agent-loginID 20001. In the sample configuration, an
Avaya one-X® Deskphone logged in using agent-loginID 20001 and the configured Password to staff and take calls for skill 1.
change agent-loginID 20001 Page 1 of 2
AGENT LOGINID
Login ID: 20001 AAS? n
Name: Agent 1 AUDIX? n
TN: 1 LWC Reception: spe
COR: 1 LWC Log External Calls? n
Coverage Path: AUDIX Name for Messaging:
Security Code:
LoginID for ISDN/SIP Display? n
Password:
Password (enter again):
Auto Answer: station
MIA Across Skills: system
ACW Agent Considered Idle: system
Aux Work Reason Code Type: system
Logout Reason Code Type: system
Maximum time agent in ACW before logout (sec): system
Forced Agent Logout Time: :
The following abridged screen shows Page 2 for agent-loginID 20001. Note that the Skill Number
(SN) has been set to 1. change agent-loginID 20001 Page 2 of 2
AGENT LOGINID
Direct Agent Skill: Service Objective? n
Call Handling Preference: skill-level Local Call Preference? n
SN RL SL SN RL SL
1: 1 1 16: 31: 46:
2: 17: 32: 47:
3: 18: 33: 48:
To enable a telephone or one-X® Agent client to log in with the agent-loginID shown above, ensure that Expert Agent Selection (EAS) Enabled is set to “y” as shown in the screen below.
change system-parameters features Page 11 of 19
FEATURE-RELATED SYSTEM PARAMETERS
CALL CENTER SYSTEM PARAMETERS
EAS
Expert Agent Selection (EAS) Enabled? y
Minimum Agent-LoginID Password Length: 4
5.10. Private Numbering
The change private-unknown-numbering command may be used to define the format of numbers sent to Verizon in SIP headers such as the “Contact” and “P-Asserted-Identity” headers.
In the bolded rows shown in the example abridged output below, entries are made for the specific
Communication Manager Vector Directory Numbers (VDN) illustrated in the prior section.
Without this configuration, calls to the VDNs would result in a 5-digit user portion of the Contact header in the 183 with SDP and 200 OK returned to Verizon. Although this did not present any
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user-perceivable problem in the sample configuration, the configuration in the bolded rows below illustrate how to cause Communication Manager to populate the Contact header with user portions that correspond with a Verizon IPCC number. In the course of the testing, multiple Verizon tollfree numbers were associated with different Communication Manager extensions and functions.
change private-numbering 0 Page 1 of 2
NUMBERING - PRIVATE FORMAT
Ext Ext Trk Private Total
Len Code Grp(s) Prefix Len
5 10 5 Total Administered: 7
5 12 5 Maximum Entries: 540
5 14 5
5 20 5
5 10001 2 8668523221 10
5 10003 2 8668510107 10
5 10004 2 8668508170 10
5.11. Incoming Call Handling Treatment for Incoming Calls
In general, the “incoming call handling treatment” for a trunk group can be used to manipulate the digits received for an incoming call if necessary. Since Session Manager is present, Session
Manager can be used to perform digit conversion, and digit manipulation via the Communication
Manager incoming call handling table is not necessary. In alternative configurations, if the toll-free number sent by Verizon was not changed before reaching Communication Manager, then the
Verizon IPCC number could be mapped to a Communication Manager extension using the incoming call handling treatment of the receiving trunk group. As an example, the following screen illustrates a conversion of toll-free number 8668502380 to extension 14000 when the call arrives on trunk group 1.
change inc-call-handling-trmt trunk-group 2 Page 1 of 30
INCOMING CALL HANDLING TREATMENT
Service/ Number Number Del Insert
Feature Len Digits
public-ntwrk 10 8668502380 10 14000
5.12. Communication Manager Stations
In the sample configuration, five digit station extensions were used with the format 120xx. Since this configuration is not unique to Verizon, a minimum of information is presented simply to assist in understanding verification traces presented in subsequent sections.
The following abbreviated screen shows an example extension for an Avaya H.323 IP telephone also used by Avaya one-X® Communicator. Call appearances and desired features (e.g., call forwarding, EC500, etc.) can be assigned to the station on page 4 (not shown).
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change station 12005 Page 1 of 5
STATION
Extension: 12005 Lock Messages? n BCC: M
Type: 9630 Security Code: * TN: 1
Port: S00024 Coverage Path 1: COR: 1
Name: IP Phone 9630-H.323 Coverage Path 2: COS: 1
Hunt-to Station:
STATION OPTIONS
Time of Day Lock Table:
Loss Group: 19 Personalized Ringing Pattern: 1
Message Lamp Ext: 12003
Speakerphone: 2-way Mute Button Enabled? y
Display Language: english Button Modules: 0
Survivable GK Node Name:
Survivable COR: internal Media Complex Ext:
Survivable Trunk Dest? y IP SoftPhone? y
IP Video Softphone? y
Short/Prefixed Registration Allowed: default
Customizable Labels? y
The following abbreviated screen shows an example extension used by an Avaya one-X® Agent client. Call appearances and appropriate features (e.g., uui-info, aux-work, etc.) can be assigned on page 4 (not shown).
change station 12004 Page 1 of 5
STATION
Extension: 12004 Lock Messages? n BCC: 0
Type: 9641 Security Code: * TN: 1
Port: S00002 Coverage Path 1: COR: 1
Name: Test Agent Coverage Path 2: COS: 1
Hunt-to Station:
STATION OPTIONS
Time of Day Lock Table:
Loss Group: 19 Personalized Ringing Pattern: 1
Message Lamp Ext: 12004
Speakerphone: 2-way Mute Button Enabled? y
Display Language: english Button Modules: 0
Survivable GK Node Name:
Survivable COR: internal Media Complex Ext:
Survivable Trunk Dest? y IP SoftPhone? y
5.13. Saving Communication Manager Configuration Changes
The command save translation all can be used to save the configuration.
save translation all
SAVE TRANSLATION
Command Completion Status Error Code
Success 0
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6. Configure Avaya Aura® Session Manager Release 6.3
This section illustrates relevant aspects of the Session Manager configuration used in the verification of these Application Notes.
Note – The following sections assume that Session Manager and System Manager have been installed and that network connectivity exists between System Manager and Session Manager.
Session Manager is managed via System Manager. Using a web browser, access “https://<ip-addr of System Manager>/SMGR”. In the Log On screen, enter appropriate User ID and Password and press the Log On button (not shown).
Once logged in, a Home Screen is displayed. An abridged Home Screen is shown below.
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Under the heading “Elements” in the center, select Routing. The right side of the screen, illustrated below, outlines a series of steps. The sub-sections that follow are in the same order as the steps outlined under Introduction to Network Routing Policy in the abridged screen shown below.
Scroll down to review additional information as shown below. In these Application Notes, all steps are illustrated with the exception of Step 9, since “Regular Expressions” were not used.
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6.1. Domains
To view or change SIP domains, select Routing Domains. Click on the checkbox next to the name of the SIP domain and Edit to edit an existing domain, or the New button to add a domain.
Click the Commit button after changes are completed.
The following screen shows a list of configured SIP domains. The Session Manager used in the verification of these Application Notes was shared among other Avaya interoperability test efforts.
The domain “avayalab.com” was used for communication with Avaya SIP Telephones and other
Avaya systems and applications. The domain “avayalab.com” is not known to the Verizon production service.
The domain “adevc.avaya.globalipcom.com” is the domain known to Verizon as the enterprise SIP domain. For example, for calls from the enterprise site to Verizon, this domain can appear in the
From and P-Asserted-Identity headers in the INVITE message sent to Verizon.
6.2. Locations
To view or change locations, select Routing Locations. The following screen shows an abridged list of configured locations. Click on the checkbox corresponding to the name of a location and Edit to edit an existing location, or the New button to add a location. Click the
Commit button (not shown) after changes are completed. Assigning unique locations can allow
Session Manager to perform location-based routing, bandwidth management, and call admission control.
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The following screen shows the location details for the location named “Vz-ASBCE”, corresponding to the Avaya SBCE relevant to these Application Notes. Later, the location with name “Vz-ASBCE” will be assigned to the corresponding Avaya SBCE SIP Entity.
The Location Pattern is used to identify call routing based on IP address. Session Manager matches the IP address of SIP Entities against the patterns defined in this section. If a call is from a
SIP Entity that does not match the IP address pattern then Session Manager uses the Location administered in the SIP Entity form. In this sample configuration Locations are added to SIP
Entities in Section 6.4, so it was not necessary to add a pattern.
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The location named “Loc19-CM” shown in the following screen will later be assigned to the corresponding Communication Manager SIP Entity. In the sample configuration, other location parameters (not shown) retained the default values.
The following screen shows the location details for the location named “SM-Denver”, corresponding to Session Manager. This location was created during the installation of Session
Manager and was assigned to the Session Manager SIP Entity. In the sample configuration, other location parameters (not shown) retained the default values.
6.3. Adaptations
To view or change adaptations, select Routing Adaptations. Click on the checkbox corresponding to the name of an adaptation and Edit to edit an existing adaptation, or the New button to add an adaptation. Click the Commit button after changes are completed (not shown).
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The following screen shows a portion of the list of adaptations that were available in the sample configuration, not all of which are applicable to these Application Notes
The adapter named “CM63-TG1-VzIPT” shown in the following screen will later be assigned to the SIP Entity linking Session Manager to Communication Manager for calls involving Verizon
Business IP Trunk service. This adaptation uses the “DigitConversionAdapter” and specifies the following parameters:
“fromto=true”. This adapts the From and To headers along with the Request-Line and PAI headers.
“osrcd=avayalab.com”. This enables the source domain to be overwritten with
“avayalab.com”. For example, for inbound PSTN calls from Verizon to Communication
Manager, the PAI header will contain “avayalab.com”.
Depending on the Communication Manager configuration, it may not be necessary for Session
Manager to adapt the domain in this fashion.
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Scrolling down, the following screen shows a portion of the “CM63-TG2-VzIPCC” adapter that can be used to convert digits between the Communication Manager extension numbers (user extensions, VDNs) and the toll-free numbers assigned by Verizon.
An example portion of the settings for “Digit Conversion for Outgoing Calls from SM” (i.e., inbound to Communication Manager) is shown below. During the testing, this digit conversion was varied to allow the same toll-free number to be used to test different Communication Manager destinations.
In general, digit conversion such as this that converts a Verizon IPCC number to a Communication
Manager extension can be performed in Communication Manager or in Session Manager. In the example screens shown above, before sending the SIP INVITE to Communication Manager,
Session Manager would adapt a dialed number of 8668510107 to the VDN 10003 associated with testing Refer with UUI. As such, it would not be necessary to use the incoming call handling table of the receiving Communication Manager trunk group to convert the toll-free number to its corresponding extension
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The adapter named “VerizonIPCC-SBC” shown below will later be assigned to the SIP Entity for the Avaya SBCE, specifying that all communication from Session Manager to the Avaya SBCEs will use this adapter.
This adaptation uses the “VerizonAdapter” module and specifies the following parameters:
“fromto=true”. This adapts the From and To headers along with the Request-Line and PAI headers.
“osrcd=adevc.avaya.globalipcom.com”. This enables the source domain to be overwritten with “adevc.avaya.globalipcom.com”. For example, for outbound PSTN calls from the
Avaya CPE to Verizon, the PAI header will contain “adevc.avaya.globalipcom.com” as expected by Verizon.
“odstd=172.30.205.55”. This enables the destination domain to be overwritten with
“172.30.205.55”, the Verizon IPCC service node IP Address.
Scrolling down, the following screen shows an abridged portion of the settings for “Digit
Conversion for Outgoing Calls from SM”. Although the direction of actual calls involving Verizon
IPCC service are “inbound” to Communication Manager, SIP headers in responses from
Communication Manager can be adapted using the “Digit Conversion for Outgoing Calls from
SM” area of the “VerizonIPCC-SBC” adaptation.
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6.4. SIP Entities
To view or change SIP entities, select Routing SIP Entities. Click the checkbox corresponding to the name of an entity and Edit to edit an existing entity, or the New button to add an entity.
Click the Commit button after changes are completed.
The following screen shows the list of configured SIP entities in the shared test environment.
The following screen shows the upper portion of the SIP Entity Details corresponding to “ASM”.
The FQDN or IP Address field for “ASM” is the Session Manager Security Module IP Address
(10.64.19.226), which is used for SIP signaling with other networked SIP entities. The Type for this SIP entity is “Session Manager”. Select an appropriate location for the Session Manager from the Location drop-down menu. In the shared test environment, the Session Manager used location
“SM-Denver”. The default SIP Link Monitoring parameters may be used. Unless changed elsewhere, links from other SIP entities to this instance of Session Manager will use the default
SIP Link Monitoring timers, configurable at the Session Manager level. If desired, these timers may be customized for each entity.
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Scrolling down, the following screen shows the middle portion of the SIP Entity Details, a listing of the Entity Links previously configured for “ASM”. The links relevant to these Application
Notes are described in the subsequent section.
Scrolling down, the following screen shows the lower portion of the SIP Entity Details, illustrating the configured ports for “ASM”. This section is only present for Session Manager SIP entities. This section defines a default set of ports that Session Manager will use to listen for SIP requests, typically from registered SIP endpoints. Session Manager can also listen on additional ports defined elsewhere such as the ports specified in the SIP Entity Link definition in Section 6.5.
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The following screen shows the upper portion of the SIP Entity Details corresponding to
“Vz_ASBCE-1”. The FQDN or IP Address field is configured with the Avaya SBCE inside IP
Address (10.64.19.140). “SIP Trunk” is selected from the Type drop-down menu. This Avaya
SBCE has been assigned to Location “Vz-ASBCE”, and the “VerizonIPCC-SBC” adapter is applied. Other parameters (not shown) retain default values.
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The following screen shows a portion of the SIP Entity Details corresponding to a
Communication Manager SIP Entity named “CM63-TG3” This is the SIP Entity that was already in place in the shared Avaya Interoperability Test Lab environment, prior to adding the Verizon IP
Trunk configuration. The FQDN or IP Address field contains the IP Address of the “processor
Ethernet” (10.64.19.155). In systems with Avaya G650 Media Gateways containing C-LAN cards,
C-LAN cards may also be used as SIP entities, instead of, or in addition to, the “processor
Ethernet”. “CM” is selected from the Type drop-down menu and “Loc19-CM” is selected for the
Location.
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The following screen shows the SIP Entity Details for an entity named “CM63-TG2”. This entity uses the same FQDN or IP Address (10.64.19.155) as the prior entity with name “CM63-TG3”; both correspond to Communication Manager processor Ethernet IP address. Later, a unique port,
5071, will be used for the Entity Link to “CM63-TG2”. Using a different port is one approach that will allow Communication Manager to distinguish traffic originally from Verizon IPCC from other
SIP traffic arriving from the same IP Address of the Session Manager, such as SIP traffic associated with SIP Telephones or other SIP-integrated applications. “CM” is selected from the
Type drop-down menu. The Adaptation “CM63-TG2-VzIPCC” is applied to this SIP entity.
Recall that this adapter is used to map the Verizon IPCC toll-free numbers to the corresponding
Communication Manager extensions. “Loc19-CM” is selected for the Location.
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6.5. Entity Links
To view or change Entity Links, select Routing Entity Links. Click on the checkbox corresponding to the name of a link and Edit to edit an existing link, or the New button to add a link. Click the Commit button after changes are completed.
The following screen shows a list of configured links. In the screen below, the links named “SM to
Vz_ASBCE-1” and “SM to CM63-TG2” are most relevant to these Application Notes. Each link uses the entity named “ASM” as SIP Entity 1, and the appropriate entity, such as “Vz_ASBCE-
1”, for SIP Entity 2. Note that there are multiple SIP Entity Links, using different TLS ports, linking the same “ASM” with the processor Ethernet of Communication Manager. For example, for one link named “SM to CM63-TG3”, both entities use TLS and port 5061. For the entity link used by Verizon IPCC named “SM to CM63-TG2”, both entities use TLS and port 5071.
The link named “SM to CM63-TG3” links Session Manager “ASM” with Communication
Manager processor Ethernet. This link existed in the configuration prior to adding the Verizon
IPCC-related configuration. This link, using port 5061, can carry traffic between Session Manager and Communication Manager that is not necessarily related to calls with Verizon, such as traffic related to SIP Telephones registered to Session Manager.
The link named “SM to CM63-TG2” also links Session Manager “ASM” with Communication
Manager processor Ethernet. However, this link uses port 5071 for both entities in the link. This link was created to allow Communication Manager to distinguish calls from Verizon IPCC from other calls that arrive from the same Session Manager. Other methods of distinguishing traffic could be used, if desired.
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6.6. Time Ranges
To view or change Time Ranges, select Routing Time Ranges. The Routing Policies shown subsequently will use the “24/7” range since time-based routing was not the focus of these
Application Notes. Click the Commit button (not shown) after changes are completed.
6.7. Routing Policies
To view or change routing policies, select Routing Policies. Click on the checkbox corresponding to the name of a policy and Edit to edit an existing policy, or New to add a policy.
Click the Commit button after changes are completed (not shown).
The following screen shows the Routing Policy Details for the policy named “To-CM63-TG2” associated with incoming toll-free calls from Verizon IPCC to Communication Manager. Observe the SIP Entity as Destination is the entity named “CM63-TG2”.
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6.8. Dial Patterns
To view or change dial patterns, select Routing Dial Patterns. Click on the checkbox corresponding to the name of a pattern and Edit to edit an existing pattern, or New to add a pattern. Click the Commit button after changes are completed.
The following screen illustrates an example dial pattern used to verify inbound PSTN calls to the enterprise. When a user on the PSTN dials a toll-free number such as 866-850-2380, Verizon delivers the number to the enterprise, and the Avaya SBCE sends the call to Session Manager. The dial pattern below matches on 866-850-2380 specifically. Dial patterns can alternatively match on ranges of numbers. Under Originating Locations and Routing Policies, the routing policy named
“To-CM63-TG2” is chosen when the call originates from Originating Location Name “Vz-
ASBCE”. This sends the call to Communication Manager using port 5071 as described previously.
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7. Configure Avaya Session Border Controller for Enterprise
Release 6.2
These Application Notes assume that the installation of the Avaya SBCE and the assignment of all
IP addresses have already been completed, including the management IP address.
In the sample configuration, the management IP is 10.80.140.140. Access the web management interface by entering https://<ip-address> where <ip-address> is the management IP address assigned during installation. Log in with the appropriate credentials. Click Log In.
The main page of the Avaya SBCE will appear.
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To view system information that was configured during installation, navigate to System
Management. A list of installed devices is shown in the right pane. In the case of the sample configuration, a single device named “VZ_1” is shown. To view the configuration of this device, click View as highlighted below.
The System Information screen shows the Network Settings, DNS Configuration, and
Management IP information provided during installation and corresponds to Figure 1. The Box
Type was set to “SIP” and the Deployment Mode was set to “Proxy”. Default values were used for all other fields. Note that the Management IP must be on a separate subnet from the IP interfaces designated for SIP traffic.
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7.1. Network Management
The Network Management screen is where the network interface settings are configured and enabled. During the installation process, certain network-specific information is defined such as device IP address(es), public IP address(es), netmask, gateway, etc., to interface the device to the network. It is this information that populates the various Network Management tab displays, which can be edited as needed to optimize device performance and network efficiency. Navigate to
Device Specific Settings Network Management and verify the IP addresses assigned to the interfaces and that the interfaces are enabled. The following screen shows the internal interface is assigned to A1 and the external interface is assigned to B1.
The following screen shows interface A1 and B1 are Enabled. To enable an interface click the corresponding Toggle State button.
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7.2. Routing Profile
Routing profiles define a specific set of packet routing criteria that are used in conjunction with other types of domain policies to identify a particular call flow and thereby ascertain which security features will be applied to those packets. Parameters defined by Routing Profiles include packet transport settings, name server addresses and resolution methods, next hop routing information, and packet transport types.
Create a Routing Profile for Session Manager and Verizon IP Trunk service. To add a routing profile, navigate to Global Profiles Routing and select Add. Enter a Profile Name and click
Next to continue.
In the shared test environment the following screen shows Routing Profile “Route to SM6.3” created for Session Manager. The Next Hop Server 1 IP address must match the IP address of
Session Manager Entity created for Avaya SBCE in Section 6.4. The Outgoing Transport is set to TCP and matched the Protocol set in the Session Manager Entity Link for Avaya SBCE in
Section 6.5.
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The following screen shows Routing Profile “Route To Vz_IPCC” created for Verizon. Enter the
IP address and port of the Verizon SIP signaling interface as Next Hop Server 1, as shown below.
It is only necessary to include the port after the IP address when it is not the default SIP port.
Choose UDP for Outgoing Transport, and click Finish.
7.3. Topology Hiding Profile
The Topology Hiding profile manages how various source, destination and routing information in
SIP and SDP message headers are substituted or changed to maintain the integrity of the network.
It hides the topology of the enterprise network from external networks.
Click the Add button (not shown) to add a new profile, or select an existing topology hiding profile to edit. If adding a profile, a screen such as the following is displayed. Enter a Profile
Name such as “Avaya” shown below. Click Next.
In the resultant screen, click the Add Header button in the upper right multiple times to reveal additional headers.
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In the Replace Action column an action of “Auto” will replace the header field with the IP address of the Avaya SBCE interface and the “Overwrite” will use the value in the Overwrite Value. In the example shown, this profile will later be applied in the direction of the Session Manager and
“Overwrite” has been selected for the To/From and Request-Line headers and the shared interop lab domain of “avayalab.com” has been inserted. Click Finish.
After configuration is completed, the Topology Hiding for profile “Avaya” will appear as follows.
This profile will later be applied to the Server Flow for Avaya.
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Similarly, create a Topology Hiding profile for Verizon. The following screen shows Topology
Hiding profile “IPCC_Topology_Hiding” created for Verizon. This profile will later be applied to the Server Flow for Verizon.
7.4. Server Interworking Profile
The Server Internetworking profile configures and manages various SIP call server-specific parameters such as TCP and UDP port assignments, heartbeat signaling parameters (for HA deployments), DoS security statistics, and trusted domains. Interworking Profile features are configured based on different Trunk Servers. There are default profiles available that may be used as is, or modified, or new profiles can be configured as described below.
In the sample configuration, separate Server Interworking Profiles were created for Avaya and
Verizon IPCC.
7.4.1 Server Interworking
– Avaya
Navigate to Global Profiles Server Interworking and click the Add Profile button (not shown) to add a new profile or select an existing interworking profile. If adding a profile, a screen such as the following is displayed. Enter an appropriate Profile Name such as “Avaya” shown below. Click Next.
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The following screens illustrate the “General” parameters used in the sample configuration for the
Interworking Profile named “Avaya”. Most parameters retain default values. In the sample configuration, RFC3264 – a=sendonly is selected and T.38 support is checked.
Click Next to advance to through both the Privacy / DTMF parameters screen, and the SIP /
Transport Timers parameters screen, which may retain default values.
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The following screen illustrates the Advanced Settings configuration. The Topology Hiding:
Change Call-ID is unchecked and the AVAYA Extensions is checked. All other parameters shown are default values. Note that the default configuration will result in Record-Route headers in
SIP messages.
7.4.2 Server Interworking
– Verizon IPCC
Click the Add button (not shown) to add a new profile or select an existing interworking profile. If adding a profile, a screen such as the following is displayed. Enter an appropriate Profile Name such as “Verizon-IPCC” shown below. Click Next.
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The following screens illustrate the “General” parameters used in the sample configuration for the
Interworking Profile named “Verizon-IPCC”. Most parameters retain default values. In the sample,
Hold Support was set for RFC3264, and all other fields retained default values.
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The following screen illustrates the Advanced Settings configuration. The Topology Hiding:
Change Call-ID and Change Max Forwards defaults were changed to “No”. All other parameters shown are default values. Note that the default configuration will result in Record-
Route headers in SIP messages.
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7.5. Server Configuration
The Server Configuration screen contains four tabs: General, Authentication, Heartbeat, and
Advanced. Together, these tabs configure and manage various SIP call server-specific parameters such as TCP and UDP port assignments, heartbeat signaling parameters, DoS security statistics, and trusted domains.
Select Global Profiles Server Configuration from the left-side menu as shown below.
7.5.1 Server Configuration for Session Manager
Click the Add button (not shown) to add a new profile, or select an existing profile to edit. If adding a profile, a screen such as the following is displayed. Enter an appropriate Profile Name such as “Avaya_SM6.3” shown below. Click Next.
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The following screens illustrate the Server Configuration for the Profile name “Avaya_SM6.3”.
On the General tab, select “Call Server” from the Server Type drop-down menu. In the IP
Addresses / Supported FQDNs area, the IP Address of the Session Manager SIP signaling interface in the sample configuration is entered. This IP Address is 10.64.19.226. In the Supported
Transports area, TCP is selected, and the TCP Port is set to 5060. This configuration corresponds with the Session Manager entity link configuration for the entity link to the Avaya
SBCE created in Section 6.4. If adding a new profile, click Next (not shown). If editing an existing profile, click Finish.
If adding the profile, click Next to accept default parameters for the Authentication tab (not shown), and advance to the Heartbeat area. If editing an existing profile, select the Heartbeat tab and click Edit (not shown).
Avaya SBCE can be configured to source “heartbeats” in the form of SIP OPTIONS. In the sample configuration, with one Session Manager, this configuration is optional.
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If Avaya SBCE-sourced OPTIONS messages are desired, check the Enable Heartbeat box. Select
“OPTIONS” from the Method drop-down menu. Select the desired frequency that the Avaya
SBCE will source OPTIONS to this server. The From URI and To URI may be filled in to configure easily identifiable URIs to appear in SIP OPTIONS sourced by the Avaya SBCE towards Session Manager. If adding a new profile, click Next (not shown). If editing an existing profile, click Finish (not shown).
If adding a profile, click Next to continue to the “Advanced” settings (not shown). If editing an existing profile, select the Advanced tab and Edit (not shown). In the resultant screen, select
Enable Grooming to allow the same TCP connection to be used for all SIP messages from this device. Select the Interworking Profile “Avaya” created previously. Click Finish.
7.5.2 Server Configuration for Verizon IPCC
Click the Add button (not shown) to add a new profile, or select an existing profile to edit. If adding a profile, a screen such as the following is displayed. Enter an appropriate Profile Name such as “IPCC_Service” shown below. Click Next.
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The following screens illustrate the Server Configuration with Profile name “IPCC_Service”. In the “General” parameters, select “Trunk Server” from the Server Type drop-down menu. In the IP
Addresses / Supported FQDNs area, the Verizon-provided IPCC service IP Address is entered.
This IP Address is 172.30.205.55. In the Supported Transports area, UDP is selected, and the
UDP Port is set to 5072. Click Next to proceed to the Authentication Tab.
If adding the profile, click Next to accept default parameters for the Authentication tab (not shown), and advance to the Heartbeat area. If editing an existing profile, select the Heartbeat tab and click Edit (not shown).
The ASBCE can be configured to source “heartbeats” in the form of SIP OPTIONS towards
Verizon. This configuration is optional. Independent of whether the ASBCE is configured to source SIP OPTIONS towards Verizon, Verizon will receive OPTIONS from the enterprise site as a result of the SIP Entity Monitoring configured for Session Manager. When Session Manager sends SIP OPTIONS to the inside private IP Address of the Avaya SBCE, the Avaya SBCE will send SIP OPTIONS to Verizon. When Verizon responds, the Avaya SBCE will pass the response to Session Manager.
If Avaya SBCE sourced OPTIONS are desired, select “OPTIONS” from the Method drop-down menu. Select the desired frequency that the SBCE will source OPTIONS. The From URI and To
URI may be filled in to configure easily identifiable URIs to appear in SIP OPTIONS sourced by the Avaya SBCE. If adding a new profile, click Next to continuing to the “Advanced” settings. If editing an existing profile, click Finish (not shown).
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If editing an existing profile, highlight the desired profile and select the Advanced tab and then click the Edit button (not shown). In the resultant screen, Enable Grooming is not used for UDP connections and left unchecked. Select the Interworking Profile “Verizon_IPCC” created previously. Click Finish.
7.6. Media Rule
Media Rules define RTP media packet parameters such as prioritizing encryption techniques and packet encryption techniques. Together these media-related parameters define a strict profile that is associated with other SIP-specific policies to determine how media packets matching these criteria will be handled by the Avaya SBCE security product.
In the sample configuration, a single media rule was created by cloning the default rule called
“default-low-med”. Select the default-low-med rule and click the Clone Rule button.
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Enter a name in the Clone Name field, such as “default-low-media-QoS” as shown below. Click
Finish.
Select the newly created rule, select the Media QoS tab and click the Edit button (not shown). In the resulting screen below, check the Media QoS Marking Enabled checkbox. Select DSCP and select “EF” for expedited forwarding as shown below. Click Finish.
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When configuration is complete, the “default-low-media-QoS” media rule Media QoS tab appears as follows.
7.7. Signaling Rule
Signaling Rules define the action to be taken (Allow, Block, Block with Response, etc.) for each type of SIP-specific signaling request and response message. When SIP signaling packets are received by Avaya SBCE, they are parsed and “pattern-matched” against the particular signaling criteria defined by these rules. Packets matching the criteria defined by the Signaling Rules are tagged for further policy matching.
To add a signaling rule, navigate to Domain Policies Signaling Rules. Click the Add button to add a new signaling rule.
In the Rule Name field, enter an appropriate name, such as “Block_Hdr_Remark” and click Next.
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In the subsequent screen (not shown), click Next to accept defaults. In the Signaling QoS screen below, select DSCP and select the desired Value for Signaling QoS from the drop-down box. In the sample configuration, “AF32” is selected for Assured Forwarding 32. Click Finish.
After this configuration, the new “Block_Hdr_Remark” will appear as follows.
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Select this rule in the center pane, then select the Request Headers tab to view the manipulations performed on the request messages such as the initial INVITE or UPDATE message. The following screen shows the “Alert-Info”, “Endpoint-View”, “P-Location" and other proprietary headers removed during the compliance test. This configuration is optional in that these headers do not cause any user-perceivable problems if presented to Verizon.
Similarly, manipulations can be performed on the SIP response messages. These can be viewed by selecting the Response Headers tab as shown below.
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7.8. Application Rule
Application Rules define which types of SIP-based Unified Communications (UC) applications the
Avaya SBCE security device will protect: voice, video, and/or Instant Messaging (IM). In addition, the maximum number of concurrent voice and video sessions the network will process can be determined in order to prevent resource exhaustion.
Select Domain Policies Application Rules from the left-side menu as shown below. In the sample configuration, a single default application rule “default-trunk” is used and will be applied to the Endpoint Policy Group in the next section.
7.9. Endpoint Policy Group
The rules created within the Domain Policy section are assigned to an Endpoint Policy Group. The
Endpoint Policy Group is then applied to a Server Flow in Section 7.12. Create a separate
Endpoint Policy Group for the enterprise and the Verizon IP Trunk. To create a new policy group, navigate to Domain Policies Endpoint Policy Groups. Select the Add button.
Enter a name in the Group Name field, such as “def_low_remark” as shown below. Click Next.
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In the sample configuration, defaults were selected for all fields, with the exception of the
Application Rule which is set to “default-trunk”, Media Rule which is set to “default-low-media-
QoS”, and the Signaling Rule, which is set to “Block_Hdr_Remark” as shown below. The selected non-default media rule and signaling rule chosen were created in previous sections. Click
Finish.
Once configuration is completed, the “default-low-remark” policy group will appear as follows.
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7.10. Media Interface
The Media Interface screen is where the SIP media ports are defined. Avaya SBCE will send SIP media on the defined ports. Create a SIP Media Interface for both the inside and outside IP interfaces.
To create a new Signaling Interface, navigate to Device Specific Settings Media Interface and click Add Media Interface. The following screen shows the media interfaces created in the sample configuration for the inside and outside IP interfaces.
7.11. Signaling Interface
The Signaling Interface screen is where the SIP signaling ports are defined. Avaya SBCE will listen for SIP requests on the defined ports. Create a Signaling Interface for both the inside and outside IP interfaces.
To create a new Signaling Interface, navigate to Device Specific Settings Signaling Interface and click Add Signaling Interface. The following screen shows the signaling interfaces created in the sample configuration for the inside and outside IP interfaces.
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7.12. End Point Flows - Server Flow
When a packet is received by Avaya SBCE, the content of the packet (IP addresses, URIs, etc.) is used to determine which flow it matches. Once the flow is determined, the flow points to a policy which contains several rules concerning processing, privileges, authentication, routing, etc. Once routing is applied and the destination endpoint is determined, the policies for this destination endpoint are applied. The context is maintained, so as to be applied to future packets in the same flow.
The following screen illustrates the flow through the Avaya SBCE to secure a SIP Trunk call.
Create a Server Flow for Session Manager and the Verizon IP Trunk. To create a Server Flow, navigate to Device Specific Settings End Point Flows. Select the Server Flows tab and click
Add Flow as shown below.
The following screen shows the flow named “Avaya SM6.3 Flow” used in the sample configuration. This flow uses the interfaces, policies, and profiles defined in previous sections.
Click Finish.
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Once again, select the Server Flows tab and click Add Flow. The following screen shows the flow named “Verizon_IP_Trunk” created in the sample configuration. This flow uses the interfaces, policies, and profiles defined in previous sections. Click Finish.
The following screen summarizes the Server Flows configured in the sample configuration.
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8. Verizon Business IPCC Services Suite Configuration
Information regarding Verizon Business IPCC Services suite offer can be found at http://www.verizonbusiness.com/products/contactcenter/ip/ or by contacting a Verizon Business sales representative.
The reference configuration described in these Application Notes was located in the Avaya
Solutions and Interoperability Test Lab. Access to the Verizon Business IPCC Services suite was via a Verizon Private IP (PIP) T1 connection. Verizon Business provided all of the necessary service provisioning.
8.1. Service Access Information
The following service access information (FQDN, IP addressing, ports, toll free numbers) was provided by Verizon for the sample configuration.
CPE (Avaya)
adevc.avaya.globalipcom.com
UDP port 5060
Verizon Network
172.30.205.55
UDP Port 5072
Toll Free
Numbers
866-850-2380
866-851-0107
866-851-2649
866-852-3221
866-850-6850
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9. Verification Steps
This section provides example verifications of the Avaya configuration with Verizon Business
Private IP (PIP) Trunk service.
9.1. Avaya Aura® Communication Manager Verifications
This section illustrates verifications from Communication Manager.
9.1.1 Example Incoming Call from PSTN via Verizon IPCC to Telephone
Incoming PSTN calls arrive from Verizon at Avaya SBCE, which sends the call to Session
Manager. Session Manager sends the call to Communication Manager. On Communication
Manager, the incoming call arrives via signaling group 2 and trunk group 2.
The following edited Communication Manager list trace tac trace output shows a call incoming on trunk group 2. The PSTN telephone dialed 866-850-6850. Session Manager mapped the number received from Verizon to the extension of a Communication Manager telephone (x12005).
Extension 12005 is an IP Telephone with IP address 10.64.19.101 in Region 1. Initially, the G450
Media Gateway (10.64.18.81) is used, but as can be seen in the final trace output, once the call is answered, the final RTP media path is “ip-direct” from the IP Telephone (10.64.19.101) to the
“inside” of the Avaya SBCE (10.64.19.140) in Region 2.
list trace tac *02 Page 1
LIST TRACE time data
/* Incoming call arrives to Communication Manager for x12005 */
10:40:01 TRACE STARTED 05/07/2013 CM Release String cold-02.0.823.0-20396
10:41:10 SIP<INVITE sip:[email protected] SIP/2.0
10:41:10 Call-ID: [email protected]
10:41:10 active trunk-group 1 member 1 cid 0x18ba
/* Communication Manager sends 183 with SDP as a result of TG 1 configuration */
10:41:10 SIP>SIP/2.0 183 Session Progress
10:41:10 Call-ID: [email protected]
10:41:10 dial 12005
10:41:10 ring station 12005 cid 0x18ba
/* G450 Gateway at 10.64.19.81, ringback tone heard by caller */
10:41:10 G711MU ss:off ps:20
rgn:1 [10.64.19.101]:3314
rgn:1 [10.64.19.81]:2050
10:41:10 G729 ss:off ps:20
rgn:2 [10.64.19.140]:35004
rgn:1 [10.64.19.81]:2054
10:41:10 xoip options: fax:T38 modem:off tty:US uid:0x50009
xoip ip: [10.64.19.81]:2054
/* User Answers call, Communication Manager sends 200 OK */
10:41:13 SIP>SIP/2.0 200 OK
10:41:13 Call-ID: [email protected]
10:41:13 active station 12005 cid 0x18ba
/* Communication Manager receives ACK to 200 OK */
10:41:13 SIP<ACK sip:[email protected]:5071;transport=tls SIP/2.0
10:41:13 Call-ID: [email protected]
<continued on next page>
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/* Communication Manager sends re-INVITE to begin shuffle to ip-direct */
10:41:13 SIP>INVITE sip:[email protected]:5060;transport=tcp
10:41:13 SIP>;gsid=ec2f2030-b734-11e2-b83f-9c8e992b0a68 SIP/2.0
10:41:13 Call-ID: [email protected]
10:41:13 SIP<SIP/2.0 100 Trying
10:41:13 Call-ID: [email protected]
/* Communication Manager receives 200 OK with SDP, sends ACK with SDP */
10:41:13 SIP<SIP/2.0 200 OK
10:41:13 Call-ID: [email protected]
10:41:13 SIP>ACK sip:[email protected]:5060;transport=tcp;gs
10:41:13 SIP>id=ec2f2030-b734-11e2-b83f-9c8e992b0a68 SIP/2.0
10:41:13 Call-ID: [email protected]
/* Final media path is ip-direct from answering IP (10.64.19.101) to inside of SBC
(10.64.19.140) */
10:41:13 G729A ss:off ps:20
rgn:2 [10.64.19.140]:35004
rgn:1 [10.64.19.101]:3314
10:41:13 G729 ss:off ps:20
rgn:1 [10.64.19.101]:3314
rgn:2 [10.64.19.140]:35004
The following screen shows Page 2 of the output of the status trunk command pertaining to this same call. Note the signaling using port 5071 between Communication Manager and Session
Manager. Note the media is “ip-direct” from the IP Telephone (10.64.19.109) to the inside IP address of Avaya SBCE (10.64.19.140) using codec G.729a.
status trunk 2/1 Page 2 of 3
CALL CONTROL SIGNALING
Near-end Signaling Loc: PROCR
Signaling IP Address Port
Near-end: 10.64.19.155 : 5071
Far-end: 10.64.19.226 : 5071
H.245 Near:
H.245 Far:
H.245 Signaling Loc: H.245 Tunneled in Q.931? no
Audio Connection Type: ip-direct Authentication Type: None
Near-end Audio Loc: Codec Type: G.729
Audio IP Address Port
Near-end: 10.64.19.101 : 3314
Far-end: 10.64.19.140 : 35006
The following screen shows Page 3 of the output of the status trunk command pertaining to this same call. Here it can be observed that G.729a codec is used.
status trunk 2/1 Page 3 of 3
SRC PORT TO DEST PORT TALKPATH src port: T00009
T00009:TX:10.64.19.140:35014/g729/20ms
S00003:RX:10.64.19.101:3314/g729a/20ms
9.1.2 Example Incoming Call Referred via Call Vector to PSTN Destination
The following edited and annotated Communication Manager list trace tac trace output shows a call incoming on trunk group 2. The call was routed to a Communication Manager vector directory
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number (VDN 10001) associated with a call vector (call vector 1). The vector answers the call, plays an announcement to the caller, and then uses a “route-to” step to cause a REFER message to be sent with a Refer-To header containing the number configured in the vector “route-to” step. The
PSTN telephone dialed 866-852-3221. Session Manager can map the number received from
Verizon to the VDN extension (x10001), or the incoming call handling table for trunk group 1 can do the same. In the trace below, Session Manager had already mapped the Verizon number to the
Communication Manager VDN extension. The annotations in the edited trace highlight key behaviors. At the conclusion, the PSTN caller that dialed the Verizon toll-free number is talking to the Referred-to PSTN destination, and no trunks (i.e., from trunk 1 handling the call) are in use.
list trace tac *02 Page 1
/* Session Manager has adapted the dialed number 8668523221 to VDN 10001 */
14:21:06 SIP<INVITE sip:[email protected] SIP/2.0
14:21:06 Call-ID: [email protected]
14:21:06 active trunk-group 1 member 1 cid 0x18dd
14:21:06 0 0 ENTERING TRACE cid 6365
14:21:06 2 1 vdn e10001 bsr appl 0 strategy 1st-found override n
14:21:06 2 1 wait 2 secs hearing ringback
14:21:06 SIP>SIP/2.0 183 Session Progress
14:21:06 Call-ID: [email protected]
14:21:06 dial 10001
14:21:06 ring vector 2 cid 0x18dd
/* Vector step plays ringback. A 183 with SDP is sent*/
14:21:06 G729 ss:off ps:20
rgn:2 [10.64.19.140]:35002
rgn:1 [10.64.19.81]:2062
14:21:06 xoip options: fax:T38 modem:off tty:US uid:0x50009
xoip ip: [10.64.19.81]:2062
14:21:08 2 2 # Play announcement to caller i...
14:21:08 2 3 announcement 11006
14:21:08 SIP>SIP/2.0 183 Session Progress
14:21:08 Call-ID: [email protected]
14:21:08 2 3 announcement: board 001V9 ann ext: 11006
/* Vector step answers call with announcement. 200 OK is sent */
14:21:08 SIP>SIP/2.0 200 OK
14:21:08 Call-ID: [email protected]
14:21:08 active announcement 11006 cid 0x18dd
14:21:08 hear annc board 001V9 ext 11006 cid 0x18dd
14:21:08 SIP<ACK sip:[email protected]:5071;transport=tls SIP/
14:21:08 SIP<2.0
14:21:08 Call-ID: [email protected]
/* Caller hears pre-REFER announcement, announcement completes, REFER sent */
14:21:11 idle announcement cid 0x18dd
14:21:11 2 4 # Refer the call to PSTN Destin...
14:21:11 2 5 route-to number ~r+13035387024 cov n if unconditionally
14:21:11 SIP>REFER sip:[email protected]:5060;transport=tcp;
14:21:11 SIP>gsid=a5787640-b753-11e2-b83f-9c8e992b0a68 SIP/2.0
14:21:11 Call-ID: [email protected]
/* Communication Manager receives 202 Accepted sent by Verizon IPCC */
14:21:11 SIP<SIP/2.0 202 Accepted
14:21:11 Call-ID: [email protected]
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/* Verizon IPCC sends re-INVITE with c=0.0.0.0 SDP and 200 OK/ACK occur */
14:21:11 SIP<INVITE sip:[email protected]:5071;transport=tls S
14:21:11 SIP<IP/2.0
14:21:11 Call-ID: [email protected]
14:21:11 SIP>SIP/2.0 100 Trying
14:21:11 Call-ID: [email protected]
14:21:11 SIP>SIP/2.0 200 OK
14:21:11 Call-ID: [email protected]
/* Verizon IPCC sends NOTIFY with sipfrag 100 Trying,CM sends 200 OK */
14:21:11 SIP<NOTIFY sip:[email protected]:5071;transport=tls S
14:21:11 SIP<IP/2.0
14:21:11 Call-ID: [email protected]
14:21:11 SIP>SIP/2.0 200 OK
14:21:11 Call-ID: [email protected]
14:21:11 SIP<ACK sip:[email protected]:5071;transport=tls SIP/
14:21:11 SIP<2.0
14:21:11 Call-ID: [email protected]
14:21:16 SIP<NOTIFY sip:[email protected]:5071;transport=tls S
14:21:16 SIP<IP/2.0
14:21:16 Call-ID: [email protected]
14:21:16 SIP>SIP/2.0 200 OK
* Note that caller does not hear ringback or any audible feedback until answer */
/* Verizon IPCC sends NOTIFY with sipfrag 200 OK and CM sends 200 OK and BYE */
14:21:16 Call-ID: [email protected]
14:21:16 2 5 LEAVING VECTOR PROCESSING cid 6365
14:21:16 SIP>BYE sip:[email protected]:5060;transport=tcp;gs
14:21:16 SIP>id=a5787640-b753-11e2-b83f-9c8e992b0a68 SIP/2.0
14:21:16 Call-ID: [email protected]
14:21:16 idle vector 0 cid 0x18dd
/* Trunks are now idle. Caller and refer-to target are connected by Verizon */
When the initial call arrived from Verizon, it used trunk member 1 from trunk group 2. In the final state when the PSTN caller is speaking with the answering agent at the Refer-To target, trunk member 1 is idle, reflecting the successful REFER.
status trunk 2
TRUNK GROUP STATUS
Member Port Service State Mtce Connected Ports
Busy
0077/001 T00041 in-service/idle no
0077/002 T00042 in-service/idle no
0077/003 T00043 in-service/idle no
0077/004 T00044 in-service/idle no
0077/005 T00045 in-service/idle no
0077/006 T00046 in-service/idle no
0077/007 T00047 in-service/idle no
0077/008 T00048 in-service/idle no
0077/009 T00049 in-service/idle no
0077/010 T00050 in-service/idle no
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9.2. Avaya Aura® System Manager and Avaya Aura® Session Manager
Verifications
This section contains verification steps that may be performed using System Manager for Session
Manager.
9.2.1 Verify SIP Entity Link Status
Log in to System Manager. Expand Elements Session Manager System Status SIP
Entity Monitoring, as shown below.
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From the list of monitored entities, select an entity of interest, such as “Vz_ASBCE-1”. Under normal operating conditions, the Link Status should be “UP” as shown in the example screen below.
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9.2.2 Call Routing Test
The Call Routing Test verifies the routing for a particular source and destination. To run the routing test, expand Elements Session Manager System Tools Call Routing Test, as shown below.
A screen such as the following is displayed.
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For example, the following shows a call routing test for an inbound toll-free call from the PSTN to the enterprise via the Avaya SBCE (10.64.19.140). Under Routing Decisions, observe that the call will route to the Communication Manager using the SIP entity named “CM63-TG2”. The digits are manipulated such that the Verizon toll-free number (i.e., 866-850-6850) is converted to a
Communication Manager extension (i.e., 14006) by the adapter assigned to the Communication
Manager entity. Scroll down to inspect the details of the Routing Decision Process if desired (not shown).
9.3. Avaya Session Border Controller for Enterprise Verification
9.3.1 Welcome Screen
The welcome screen shows alarms, incidents, and the status of all managed Avaya SBCEs at a glance.
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9.3.2 Alarms
A list of the most recent alarms can be found under the Alarm tab on the top left bar.
Alarm Viewer.
9.3.3 Incidents
A list of all recent incidents can be found under the incidents tab at the top left next to the Alarms.
Incident Viewer.
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Further Information can be obtained by clicking on an incident in the incident viewer.
9.3.4 Diagnostics
The full diagnostics check that can be run can run line checks in both directions.
Click on Diagnostics on the top bar, select the Avaya SBCE from the list of devices and then click
“Start Diagnostics”
A green check mark or a red x will indicate success or failure.
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9.3.5 Tracing
To take a call trace, Select Device Specific Settings Troubleshooting Tracing from the left-side menu (not shown).
Select the Packet Capture tab and set the desired configuration for a call trace and click Start
Capture.
When tracing has reached the desired number of packets the trace will stop automatically, or alternatively, click the Stop Capture button at the bottom.
Select the Captures tab at the top and the capture will be listed; select the File Name and choose to open it with an application like Wireshark.
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10. Conclusion
As illustrated in these Application Notes, Avaya Aura® Communication Manager 6.3, Avaya
Aura® Session Manager 6.3, and Avaya Session Border Controller for Enterprise can be configured to interoperate successfully with Verizon Business IP Contact Center Services suite.
This solution enables inbound toll free calls over a Verizon Business VoIP Inbound SIP trunk service connection. In addition, these Application Notes further demonstrate that the Avaya
Aura® Communication Manager implementation of SIP Network Call Redirection (SIP-NCR) can work in conjunction with Verizon's Business IP Contact Center services implementation of SIP-
NCR to support call redirection over SIP trunks inclusive of passing User-User Information (UUI).
Please note that the sample configurations shown in these Application Notes are intended to provide configuration guidance to supplement other Avaya product documentation.
11. Additional References
11.1. Avaya
Avaya product documentation, including the following, is available at http://support.avaya.com
[1] Implementing Avaya Aura® Communication Manager, Doc ID 03-603558, Release 6.3
[2] Administering Avaya Aura® Communication Manager, Doc ID 03-300509, Release 6.3
[3] Implementing Avaya Aura® Session Manager, Release 6.3
[4] Installing Service Packs for Avaya Aura® Session Manager, Release 6.3
[5] Upgrading Avaya Aura® Session Manager, Release 6.3
[6] Maintaining and Troubleshooting Avaya Aura® Session Manager, Release 6.3
[7] Implementing Avaya Aura® System Manager, Release 6.3
[8] Installing Avaya Session Border Controller for Enterprise, Release 6.2
[9] Administering Avaya Session Border Controller for Enterprise, Release 6.2
Avaya Application Notes, including the following, are also available at http://support.avaya.com
The following Application Notes cover Session Manager 6.2 with Verizon Business IP Toll Free
VoIP Inbound Service.
[VZ-IPTF] – Application Notes for Avaya Aura® Communication Manager 6.2, Avaya Aura®
Session Manager 6.2, and Avaya Session Border Controller for Enterprise with Verizon Business
IP Toll Free VoIP Inbound – Issue 1.1
The following Application Notes cover Session Manager 6.2 with Verizon Business IP Contact
Center IP-IVR Service.
[VZ-IP-IVR] – Application Notes for Avaya Aura® Communication Manager 6.2, Avaya Aura®
Session Manager 6.2, and Avaya Session Border Controller for Enterprise with Verizon Business
IP Contact Center IP-IVR – Issue 1.1
11.2. Verizon Business
The following documents may be obtained by contacting a Verizon Business Account
Representative.
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Retail VoIP Interoperability Test Plan
Network Interface Specification Retail VoIP Trunk Interface (for non-registering devices)
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Appendix A
This section covers the configuration settings of Avaya SBCE, Session Manager, and a sample endpoint as used for Remote Workers during compliance testing. In the test environment, a dedicated Avaya SBCE with private IP addresses was used to access the Verizon Business IP
Contact Center (IPCC) Services suite. To allow remote SIP endpoints access to the test environment through a public network, a separate Avaya SBCE with public IP addresses was utilized. The settings presented here simply illustrate the sample configuration used during compliance testing with Verizon IPCC, and are not intended to be prescriptive. Other routing criteria and policies may be appropriate based on different customer needs.
Standard and Advanced Session Licenses are required for Remote Worker. Contact an authorized
Avaya representative for assistance if additional licensing is required.
The following screen shows the Network Management of the Avaya SBCE. The internal interface is assigned to A1 and the external interfaces are assigned to B1. Avaya SIP endpoints registered to IP address “192.168.62.92” and retrieved firmware and configuration data from IP address “192.168.63.123”. For security reasons, any actual public IP addresses used in the configuration have been replaced with private IP addresses.
Note – A SIP Entity in Session Manager was not configured for the Avaya SBCE’s internal IP address used for Remote Worker. This keeps the interface untrusted in Session Manager, thereby allowing Session Manager to properly challenge user registration requests.
It is possible to deploy Remote Worker using the same Avaya SBCE as the one used for SIP trunking. However, separate IP addresses are needed for each function. This allows the Avaya
SBCE to enforce proper security policies as if it were two different Avaya SBCEs. Only two network interfaces on the Avaya SBCE may be active at one time, so this requires all external IP addresses to be on the same subnet so they may be applied to the same network interface.
Similarly, additional internal IP addresses must be on the same internal subnet.
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Interfaces A1 and B1 were both set to Enabled.
The following screen shows the Media Interface settings. Media interfaces were created for the inside and outside IP interfaces used for Remote Worker SIP traffic.
The following screen shows the Signaling Interface settings. Signaling interfaces were also created for the inside and outside IP interfaces used for Remote Worker SIP traffic. The interface named “Sig_Outside_92” supports TCP and TLS, while the interface named “Sig_Inside_200” supports TLS only.
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Routing Profile “SessionMgr-TLS” was created for Session Manager as shown in the following screens.
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The following screens illustrate the Server Configuration for the Profile named “SM6.3” created for Session Manager. The Authentication and Heartbeat tabs were kept at the default disabled setting.
On the Advanced tab, default profiles were specified that applies to traffic between the Avaya
SBCE and Session Manager as shown below.
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User Agents were created for each type of endpoint tested. This allows for different policies to be applied based on the type of device. For example, Avaya one-X Deskphones will use TLS and
SRTP, while one-X® Communicator and Avaya Flare® will use TCP and RTP.
The following abridged output of Session Manager’s traceSM command shows the details of an
Invite from an Avaya one-X Deskphone. The User-Agent shown in this trace will match “one-X
Deskphone” shown above with a Regular Expression of “Avaya one-X Deskphone.*”. In this expression, “.*” will match any software version listed after the user agent name.
INVITE sip:[email protected] SIP/2.0
From: <sip:[email protected]>;tag=-76557dff51bb3900-5c89896d_F1400610.80.150.111
To: <sip:[email protected]>
CSeq: 357 INVITE
Call-ID: [email protected]
Contact: <sip:[email protected]:5061;transport=tls;subid_ipcs=448140782>;+avaya-cm-line=1
Record-Route: <sip:10.64.19.200:5061;ipcs-line=930;lr;transport=tls>
Record-Route: <sips:205.168.62.92:5061;ipcs-line=930;lr;transport=tls>
Allow:
INVITE,ACK,BYE,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE,REFER,INFO,PRACK,PUBLISH,UPDATE
Supported: 100rel,eventlist,feature-ref,replaces
User-Agent: Avaya one-X Deskphone 6.2.2.17 (40235)
Max-Forwards: 69
Via: SIP/2.0/TLS 10.64.19.200:5061;branch=z9hG4bK-s1632-001744755540-1--s1632-
Via: SIP/2.0/TLS 10.80.150.111:5061;branch=z9hG4bK165_51bb39027017c28d-5c899bb5_I14006
Accept-Language: en
Content-Type: application/sdp
Content-Length: 416
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Relay Services are used to define how firmware updates and configuration data are routed for remote endpoints. The following screen shows the Application Relay tab with the two application relays created for the sample configuration. This allows for both HTTP and HTTPS traffic to be routed to the appropriate internal file server. The Remote IP:Port was set to the IP address and port of the internal file server used to provide the firmware updates and configuration data for the remote endpoints. The Listen IP:Port was set to the IP address and port of the Avaya SBCE’s external IP address designated for file transfers. The Connected IP was set to the internal IP address of the Avaya SBCE.
A Cluster Proxy is used for Personal Profile Manager (PPM) data and Presence services. The following screen shows the cluster proxy “remotephones” created in the sample configuration. A
Presence Services server was not part of the sample configuration. Therefore, configuration of the cluster proxy for use with Presence is not shown and only configuration related to PPM data is present.
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On the Primary tab, PPM traffic received on Device IP “192.168.62.92” will be routed to the
Configuration Server Client Address “10.64.19.200”. The Real IP field is not used for PPM, so any IP address can be entered, e.g., “1.2.3.4”. This enables the remote Avaya SIP endpoints to send and receive PPM information to and from Session Manager via the Avaya SBCE.
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The following screens show the Application Rules “MaxVoiceSession” and “remotephones” used in the sample configuration. In an actual customer installation, set the Maximum Concurrent
Sessions for the Voice application to a value slightly larger than the licensed sessions. For example, if licensed for 300 session set the values to “500”. For the “MaxVoiceSession” rule, the
Maximum Session Per Endpoint matches the Maximum Concurrent Sessions. For the application rule applied to the Remote Workers subscriber flows, a value of “10” is recommended for the Maximum Session Per Endpoint.
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The following screens show Media Rules “New-Avaya-Enc” and “default-low-med” that will later be assigned to the End Point Policy Groups. Note that both rules have Interworking checked.
Based on how calls are routed through Avaya SBCE, this will convert SRTP media to RTP and vice versa. In the sample configuration, Avaya SBCE will convert the SRTP media stream from remote Avaya 96x1 SIP Telephones to RTP towards the enterprise and also towards remote endpoints using TCP. Avaya SBCE will also convert RTP media from calls originating from
Session Manager to SRTP towards Avaya 96x1 SIP Telephones using TLS through the external IP interface. The “New-Avaya-Enc” policy was cloned from the existing “default-low-med” policy.
The parameters on the other tabs not shown retained their default values.
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The following End Point Policy Groups will later be assigned to the subscriber and server flows.
The “Remote_User_SRTP” policy uses the “remotephones” Application rule and the “New-
Avaya-Enc” Media rule.
The “Remote_User” policy uses the “remotephones” Application rule and the “default-low-med”
Media rule.
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The “SM” policy uses the “MaxVoiceSession” Application rule and the “default-low-med” Media rule.
Three Subscriber Flows were created for Remote Workers. One for each User Agent previously created.
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The following screen shows the details of the flow named “Remote-User-96x1” used in the sample configuration. This flow will match traffic from remote Avaya 96x1 Series IP Telephones set to use TLS. Note that the User Agent was set to “one-X Deskphone” and that the End Point Policy
Group was set to “Remote_User_SRTP”.
The “Remote-User-one-X” flow will match traffic from remote one-X® Communicator devices set to use TCP. Note that the User Agent was set to “one-X Communicator” and that the End Point
Policy Group was set to “Remote_User”.
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The “Remote-User-Flare” flow will match traffic from remote Avaya Flare® devices set to use
TCP. Note that the User Agent was set to “Flare” and that the End Point Policy Group was set to
“Remote_User”.
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The following screens show the Server Flows settings for Session Manager.
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In the sample configuration, the internal IP address of the Avaya SBCE used for Remote Worker was added to Session Manager’s SIP Firewall Whitelist and PPM limiting was disabled.
To add an IP address to the Whitelist, log into System Manager and navigate to Session Manager
Network Configuration SIP Firewall. Select the Session Manager listed in the top section and click Edit SM Default. Select the Whitelist tab towards the bottom of the screen and click
New. Enter the internal IP address of Avaya SBCE used for Remote Workers in the Value field and “255.255.255.255” in the Mask field. Click Apply As Current to save the configuration.
To disable PPM limiting, navigate to Session Manager Session Manager Administration in the left-hand navigation pane and click View (not shown). A screen such as the following is displayed.
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Scroll down to the Personal Profile manager (PPM) – Connection Settings. Uncheck Limited
PPM Client Connections and PPM Packet Rate Limiting.
The following screens show an Avaya one-X® Deskphone SIP Emulator illustrating the administration settings of a SIP endpoint used for Remote Worker. Note that the HTTPS File
Server is set to the external IP address of the Avaya SBCE designated for firmware and configuration file transfers. Under SIP Global Settings, the SIP Domain is set to “avayalab.com”.
The domain expected by Session Manager.
Under SIP Proxy Settings, the SIP Proxy Server is set to the external IP address of Avaya SBCE designated for Remote Worker SIP traffic. The Transport Type and SIP Port should be set according to device type. For example, “TLS” and “5061” for one-X® Deskphones, and “TCP” and “5060” for one-X® Communicator and Flare® Experience.
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©
2014 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
DevConnect Program at [email protected]
.
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