Application Notes for Avaya Aura Communication Manager and Acme Packet Net-Net Session Director with Sprint SIP Trunking Service Issue 1.

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1 Avaya Solution & Interoperability Test Lab Application Notes for Avaya Aura Communication Manager and Acme Packet Net-Net Session Director with Sprint SIP Trunking Service Issue 1.0 Abstract These Application Notes describe the steps required to configure Session Initiation Protocol (SIP) Trunking between Sprint SIP Trunking service and an Avaya SIP- enterprise solution. The Avaya solution consists of Avaya Aura Communication Manager and an Acme Packet 3800 Net-Net Session Border Controller, along with various Avaya endpoints. Sprint is a member of the Avaya DevConnect Service Provider program. Information in these Application Notes has been obtained through compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab. 1 of 31

2 1. Introduction These Application Notes describe the steps required to configure Session Initiation Protocol (SIP) Trunking between Sprint SIP Trunking service and an Avaya SIP- enterprise solution. The Avaya solution consists of Avaya Aura Communication Manager and an Acme Packet 3800 Net-Net Session Border Controller (SBC), along with various Avaya endpoints. Customers using this Avaya SIP- enterprise solution with Sprint SIP Trunking are able to place and receive PSTN calls via a broadband IP WAN connection and the SIP protocol. This converged network solution is an alternative to traditional PSTN trunks such as ISDN-PRI. 2. General Test Approach and Test Results The general test approach was to connect a simulated enterprise site to the Sprint SIP Trunking service via the public Internet and exercise the features and functionality listed in Section 2.1. The simulated enterprise site was comprised of Avaya Aura Communication Manager, the Acme Packet 3800 Net-Net SBC, and various Avaya endpoints Interoperability Compliance Testing To verify SIP trunking interoperability, the following features and functionality were covered during the interoperability compliance test: Response to SIP OPTIONS queries. Incoming PSTN calls to various phone types. Phone types included H.323, digital, and analog telephones at the enterprise. Inbound PSTN calls were routed to the enterprise across the SIP trunk from the service provider. Outgoing PSTN calls from various phone types. Phone types included H.323, digital, and analog telephones at the enterprise. Outbound PSTN calls were routed from the enterprise across the SIP trunk to the service provider. Inbound and outbound PSTN calls to/from Avaya one-x Communicator (soft client). Avaya one-x Communicator supports two modes (Road Warrior and Telecommuter). Each supported mode was tested. Avaya one-x Communicator also supports two Voice Over IP (VoIP) protocols: H.323 and SIP. Only the H.323 version of Communicator was tested. Various call types including: local, long distance, international, outbound toll-free, and operator (0). Codecs G.711MU and G.729A were tested. DTMF transmission using RFC Caller ID presentation and Caller ID restriction. Response to incomplete call attempts and trunk errors. Voic navigation for inbound and outbound calls. User features such as hold and resume, call forwarding, transfer and conference. Off-net call forwarding and EC500 mobility (extension to cellular). Items not supported or not tested included the following: Inbound toll-free and emergency calls are supported by Sprint but were not tested. 2 of 31

3 2.2. Test Results Interoperability testing of Sprint SIP Trunking was completed with successful results for all test cases with the exception of the observations/limitations described below. Caller ID Block: When Caller ID Block was invoked, the same 10 digit number was being displayed for all calls that requested privacy. This was determined by Sprint to be a condition of the lab setup, and should not adversely affect service in production Support Sprint customers may obtain information for Sprint SIP Trunk service by going to or for technical support, contact a Sprint Customer Care representative. Avaya customers may obtain documentation and support for Avaya products by visiting In the United States, (866) GO-AVAYA ( ) provides access to overall sales and service support menus. Customers may also use specific numbers (provided on to directly access specific support and consultation services based upon their Avaya support agreements. 3. Reference Configuration The reference configuration used in these Application Notes is shown in Figure 1 and consists of several components: Avaya Aura Communication Manager provides the enterprise voice communication services. In this sample configuration, Avaya Aura Communication Manager runs on an Avaya S8730 Server. This solution is extensible to other Avaya S8xxx Servers as well. The Avaya Media Gateway provides the physical interfaces and resources for Avaya Aura Communication Manager. In the reference configuration, an Avaya G650 Media Gateway is used. This solution is extensible to other Avaya Media Gateways. Avaya desk phones are represented with Avaya analog, 2420 Digital, 4621 SW IP and 9630 Series IP Telephones running H.323 software, as well as an Avaya one-x Communicator softphone. An existing Avaya Modular Messaging system provides the corporate voice messaging capabilities in the reference configuration. The provisioning of Modular Messaging is beyond the scope of this document. Inbound calls were placed from the PSTN via the Sprint SIP Trunk service, to the Acme Packet Net-Net Session Director, which routed the call to Avaya Aura Communication Manager. Avaya Aura Communication Manager terminated the call to the appropriate phone or fax extension. The H.323 phones on the enterprise side registered to the Avaya Aura Communication Manager CLAN IP address on the Avaya G650 media gateway. 3 of 31

4 Figure 1: Reference configuration 3.1. Illustrative Configuration Information The specific values listed in Table 1 below and in subsequent sections are used in the reference configuration described in these Application Notes and are for illustrative purposes only. Customers must obtain and use the appropriate values for their own specific configurations. Note - The Sprint SIP Trunk service Session Border Controller IP address and DID numbers, (destination digits specified in the SIP Request URIs sent by the Sprint SIP Trunk service) are shown in this document as examples. Sprint will provide actual IP addresses and DID numbers as part of the SIP Trunk service provisioning process. Component Illustrative Value in these Application Notes Acme Packet Net-Net Session Director IP Address Avaya Aura Communication Manager C-LAN IP Address Phone Extensions 4056, 4059, 4524, 4525, 4526 Voice Messaging Pilot Extension 7999 Avaya Modular Messaging Messaging Application Server (MAS) IP Address Message Store Server (MSS) IP Address Sprint SIP Trunk Service 4 of 31

5 Component Illustrative Value in these Application Notes Acme Packet SBC IP Address DIDs Assigned by Sprint Table 1: Illustrative Values Used in these Application Notes 4. Equipment and Software Validated The following equipment and software was used for the reference configuration described in these Application Notes. Component Avaya S8730 Server Avaya G650 Media Gateway Control-LAN TN799DP Val-Announcement TN2501AP IP Media Processor TN2602AP IP Media Processor TN2302AP Digital Line TN2224 Analog Line TN746B Avaya 9630 IP Telephone Version Avaya Aura Communication Manager SP10 (R015x ) HW01 FW024 HW03 FW018 HW08 FW044 HW18 FW110 Vintage Vintage H.323 Version S3.110b (ha96xxua3_11.bin) SP Avaya one-x Communicator Avaya 4621SW IP Telephone H323 Version (a10d01b2_9_1.bin) Avaya Analog phone N/A Fax device Okidata Okifax Acme Packet 3800 Net-Net Session Director SCX6.2.0 M5P5 Sprint SIP Trunk Service Sprint SIP Trunk Service Acme Packet Net- SCX6.2.0 Net Session Director Table 2: Equipment and Software Versions 5 of 31

6 5. Avaya Aura Communication Manager This section describes the administration steps for Avaya Aura Communication Manager in support of SIP trunking integration with the Sprint SIP Trunk service. The steps are performed from the Communication Manager System Access Terminal (SAT) interface. These Application Notes assume that basic Communication Manager Administration, including stations, C-LAN and Media Processor boards, etc., has already been performed. Refer to [1], [2], and [3] for further details if necessary. Note In the following sections, only the highlighted parameters are applicable to these Application Notes. Other parameters shown should be considered informational System Parameters This section reviews the Communication Manager licenses and features that are required for the sample configuration described in these Application Notes. For required licenses that are not in the steps that follow, contact an authorized Avaya account representative to obtain the licenses. 1. Enter the display system-parameters customer-options command. On Page 2 of the system-parameters customer-options form, verify that the Maximum Administered SIP Trunks number is sufficient for the number of expected SIP trunks. display system-parameters customer-options Page 2 of 10 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP econs: 0 0 Max Concur Registered Unauthenticated H.323 Stations: 0 0 Maximum Video Capable H.323 Stations: 0 0 Maximum Video Capable IP Softphones: 0 0 Maximum Administered SIP Trunks: Maximum Administered Ad-hoc Video Conferencing Ports: 0 0 Maximum Number of DS1 Boards with Echo Cancellation: 0 0 Maximum TN2501 VAL Boards: Maximum Media Gateway VAL Sources: 0 0 Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: 0 0 (NOTE: You must logoff & login to effect the permission changes.) Figure 2: System-Parameters Customer-Options Form Page 2 6 of 31

7 2. On Page 4 of the system-parameters customer-options form, verify that the bolded fields in the following screenshot are set to y display system-parameters customer-options Page 4 of 10 OPTIONAL FEATURES Emergency Access to Attendant? y Enable dadmin Login? y Enhanced Conferencing? y Enhanced EC500? y Enterprise Survivable Server? n Enterprise Wide Licensing? n ESS Administration? n Extended Cvg/Fwd Admin? n External Device Alarm Admin? n Five Port Networks Max Per MCC? n Flexible Billing? n Forced Entry of Account Codes? n Global Call Classification? n Hospitality (Basic)? y Hospitality (G3V3 Enhancements)? n IP Trunks? y IP Stations? y ISDN Feature Plus? n ISDN/SIP Network Call Redirection? y ISDN-BRI Trunks? n ISDN-PRI? y Local Survivable Processor? n Malicious Call Trace? y Media Encryption Over IP? n Mode Code for Centralized Voice Mail? n Multifrequency Signaling? y Multimedia Call Handling (Basic)? n Multimedia Call Handling (Enhanced)? n Multimedia IP SIP Trunking? n IP Attendant Consoles? N (NOTE: You must logoff & login to effect the permission changes.) Figure 3: System-Parameters Customer-Options Form Page Dial Plan This section briefly describes the dial plan requirements and feature access codes for the sample configuration described in these Application Notes. Enter the change dialplan analysis command to provision the dial plan as shown below. 3-digit dial access codes (indicated with a Call Type of dac ) beginning with the digit 1 Trunk Access Codes (TACs) defined for trunk groups in this sample configuration conform to this format. 4-digit extensions with a Call Type of ext beginning with the digits 4 or 7 Local extensions for Communication Manager phones, agents, voic access, etc. in this sample configuration conform to this format. Feature access codes beginning with * or # to access various features available on the system, and an entry for a feature access code of 9 to access an outside line. 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 3 dac 4 4 ext 7 4 ext 9 1 fac * 3 fac # 3 fac Figure 4: Dialplan Analysis Form 7 of 31

8 5.3. IP Network Parameters For simplicity in this sample configuration, all Communication Manager elements, e.g., stations, C-LAN and MedPro boards, etc., within the Avaya enterprise site are assigned to a single IP network region and all internal calls use a single IP codec set. This section describes the steps required for administering IP network regions and IP codec sets, for both the enterprise and the Sprint SIP Trunk service. 1. Enter the change ip-codec-set x command, where x is the number of an unused IP codec set. This IP codec set will be used only for internal calls. Configure as follows: change ip-codec-set 1 Page 1 of 2 Codec Set: 1 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : G.729A n : Figure 5: IP-Codec-Set Form for Internal Calls Page 1 On Page 2 of the ip-codec-set form, set FAX Mode to t.38-standard. change ip-codec-set 1 Page 2 of 2 IP Codec Set Allow Direct-IP Multimedia? n Mode Redundancy FAX t.38-standard 0 Modem off 0 TDD/TTY US 3 Clear-channel n 0 Figure 6: IP-Codec-Set Form for Internal Calls Page 2 2. Enter the change ip-codec-set x command, where x is the number of an unused IP codec set. This IP codec set will be used for inbound Sprint SIP Trunk calls. On Page 1 of the ip-codec-set form, provision the codecs in the order shown below. change ip-codec-set 3 Page 1 of 2 Codec Set: 3 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.729A n : G.711MU n 3 20 Figure 7: IP-Codec-Set Form for Inbound Calls Page 1 8 of 31

9 On Page 2 of the ip-codec-set form, set FAX Mode to t.38-standard. change ip-codec-set 3 Page 2 of 2 IP Codec Set Allow Direct-IP Multimedia? n Mode Redundancy FAX t.38-standard 0 Modem off 0 TDD/TTY US 3 Clear-channel n 0 Figure 8: IP-Codec-Set Form for Inbound Calls Page 2 3. Enter the change ip-network-region x, where x is the number of an unused IP network region. This IP network region will be used to represent the Sprint SIP Trunk Service. On Page 1 of the ip-network-region form, set the parameters as shown below: Set the Authoritative Domain field to the IP address or FQDN of the Sprint SIP trunking SBC. For the compliance test this field was set to the IP address of the Sprint SIP trunking SBC, Codec Set Set to the codec set configured in Step 2 Intra and Inter-region IP-IP Direct Audio Set both to yes to support audio shuffling which allows the endpoints to communicate directly with each other. change ip-network-region 3 Page 1 of 19 IP NETWORK REGION Region: 3 Location: Authoritative Domain: Name: Sprint SIPT MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 3 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 3329 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 Figure 9: IP-Network-Region Form for Sprint SIP Trunk service Page 1 9 of 31

10 On Page 3 of the ip-network-region form, provision the following: codec set Set to the codec set configured in Step 2 direct WAN Set to y change ip-network-region 3 Page 3 of 19 Source Region: 3 Inter Network Region Connection Management dst codec direct WAN-BW-limits Video Intervening Dyn rgn set WAN Units Total Norm Prio Shr Regions CAC IGAR AGL 1 3 y NoLimit n all all Figure 10: IP-Network-Region Form for Sprint SIP Trunk service Page 3 4. Enter the change ip-network-region x, where x is the number of an IP network region administered for local Communication Manager Elements within the Avaya site. On Page 1 of the ip-network-region form: Set the Authoritative Domain field to the domain name configured for the enterprise. The domain of avayalab2.com was used for the compliance test. Codec Set Set to the codec set configured in Step 1 Set Intra-region and Inter-region IP-IP Direct Audio to yes to support audio shuffling which allows the endpoints to communicate directly with each other. change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: avayalab2.com Name: Enterprise MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 3329 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 Figure 11: IP-Network-Region Form for Local Communication Manager Elements Page 1 10 of 31

11 On Page 3 of the ip-network-region form, provision the following: codec set for dst rgn 1 Set to the codec set administered in Step 1 codec set for dst rgn 3 Set to the codec set administered in Step 2 direct WAN Set to y change ip-network-region 1 Page 3 of 19 Source Region: 1 Inter Network Region Connection Management dst codec direct WAN-BW-limits Video Intervening Dyn rgn set WAN Units Total Norm Prio Shr Regions CAC IGAR AGL 1 1 y NoLimit n all y NoLimit n all Figure 12: IP-Network-Region Form for Local Communication Manager Elements Page 3 5. Enter the list node-names all command, and note the node names and IP addresses of the C-LAN board for the G650 gateway that connects to Communication Manager, and the Acme Packet Net-Net Session Director. list node-names all NODE NAMES Type Name IP Address IP Acme IP Gateway IP MM IP clan IP default IP medpro IP medpro IP procr... IP ses Figure 13: Node-Names Form 11 of 31

12 5.4. Signaling Group and SIP Trunk In this sample configuration, since the Sprint SIP Trunk service sends 10-digits on inbound calls, the administration steps that follow in this section reflect that requirement. In an actual deployment, this may be different Signaling Group This section describes the steps for administering the routing of inbound calls from the PSTN. 1. Enter the add signaling-group x command, where x is the number of an unused signaling group and configure as follows: Group Type Set to sip Transport Method Set to tcp for compliance testing. For security reasons it is recommended to use TLS for deployment in production environments. Near-end Node Name - Set to the CLAN node name configured in Section 5.3, Step 5 Far-end Node Name Set to the Acme Packet node name configured in Section 5.3, Step 5 Near-end and Far-end Listen Ports are set to the default value of 5060 for Transport Method of tcp Far-end Domain Set to the IP address or FQDN of the Sprint SIP trunking SBC. DTMF over IP - Set to rtp-payload to enable Avaya Aura Communication Manager to use DTMF according to RFC 2833 Direct IP-IP Audio Connections Set to y, allowing endpoints to communicate directly to reduce the use of media processing resources when possible add signaling-group 14 Group Number: 14 IMS Enabled? n SIGNALING GROUP Group Type: sip Transport Method: tcp Near-end Node Name: clan1 Far-end Node Name: Acme Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 3 Far-end Domain: Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? n Direct IP-IP Early Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6 Figure 14: Signaling-Group Form for Sprint SIP Trunk Calls 12 of 31

13 2. Enter the add trunk-group x command, where x is the number of an unused trunk group and configure as follows: On page 1 of the add trunk-group configure the following settings: Direction Set to two-way. This allows a single trunk to be used for inbound calls and calls that need to be redirected back out of the enterprise to the PSTN. TAC Set to any value as per the dial plan Service Type Set to public-ntwrk Signaling Group Set to the number of the signaling group administered in Step 1 Number of Members Set to a value large enough to accommodate the call volume. For the compliance testing, this value was set to 14. add trunk-group 8 Page 1 of 21 TRUNK GROUP Group Number: 8 Group Type: sip CDR Reports: y Group Name: Sprint SIPT COR: 1 TN: 1 TAC: 108 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? n Figure 15: Trunk-Group Form for Sprint SIP Trunk Calls Page 1 Signaling Group: 14 Number of Members: 14 On page 3 of the add trunk-group form, configure the following settings: Set the Numbering Format field to public Set the Replace Restricted Numbers and Replace Unavailable Numbers fields to y add trunk-group 8 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: Maintenance Tests? y Numbering Format: public UUI Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y Show ANSWERED BY on Display? Y Figure 16: Trunk-Group Form for Sprint SIP Trunk Calls Page 3 On page 4 of the add trunk-group form, configure the following settings: Set Mark Users as Phone to y Set Network Call Redirection to y. This enables the use of the SIP REFER method for redirected calls such as call transfers. Set Send Diversion Header to y so that Diversion headers are sent within the SIP messaging Set Support Request History to n. This is necessary to support the use of the SIP REFER method for redirected calls such as call transfers. 13 of 31

14 change trunk-group 8 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? y Prepend '+' to Calling Number? n Send Transferring Party Information? n Network Call Redirection? y Send Diversion Header? y Support Request History? n Telephone Event Payload Type: 101 Figure 17: Trunk-Group Form for Sprint SIP Trunk Calls Page 4 3. Enter the change route-pattern x command, where x is the number of an unused route pattern. On page 1 of the change route-pattern form, provision the following: Enter a descriptive name in the Pattern Name field. Enter the Trunk Group number of the primary SIP Trunk that was configured in step 2 above in the Grp No field. For the compliance test Trunk Group 8 was used. Last, set the LAR field to next. change route-pattern 8 Page 1 of 3 Pattern Number: 8 Pattern Name: Sprint SIPT SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 8 0 n user 2: n user 3: n user 4: n user 5: n user 6: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest next 2: y y y y y n n rest 3: y y y y y n n rest 4: y y y y y n n rest Figure 18: Route Pattern Form 4. Enter the change inc-call-handling-trmt trunk-group x command, where x is the number of the trunk group administered in Step 2 above. In the inc-call-handling-trmt trunk-group form, provision an entry for each assigned DID number as follows: Called Len Enter the total number of digits for the DID number Called Number Enter enough leading digits to uniquely match the DID range, or enter the exact matching DID number. Del and Insert If necessary, enter the number of leading digits that need to be deleted from the DID number, and the specific leading digits that need to be prefixed to the DID number (after any deletion is performed), respectively, in order to match a local Communication Manager extension range. In this sample configuration, Sprint SIP Trunk service sends a 10-digit number in the SIP INVITE ( ) for an inbound call. Thus, the entry in the figure below matches this number and deletes all 10 digits, then inserts 4526 to match the local Communication Manager 14 of 31

15 extension. Provision as many entries as necessary to cover all expected DID numbers sent by the Sprint SIP Trunk service. DID numbers can map to VDNs, stations or other Communication Manager elements. change inc-call-handling-trmt trunk-group 8 Page 1 of 30 INCOMING CALL HANDLING TREATMENT Service/ Called Called Del Insert Feature Len Number public-ntwrk Figure 19: Inc-Call-Handling-Trmt Trunk-Group Form for Sprint SIP Trunk Calls Calling Party Number The calling party numbers sent on outbound calls can be specified in the public-unknownnumbering form. These numbers are displayed for Caller ID for any connected calls or redirected outbound calls. Configure an entry in the public-unknown-numbering form as follows: Ext Len Enter the total number of digits in the local extension range. Ext Code Enter enough leading digits to identify the local extension range, or enter the exact extension number. Trk Grp(s) Enter the number of the trunk group administered in Section 5.4.1, Step 2. CPN Prefix If necessary, enter enough prefix digits to form the desired connected party number, or enter the exact number that should be displayed. CPN Len Enter the total length of the number to be sent. Provision as many entries as necessary to cover all local Extension ranges. Change public-unknown-numbering 4 Page 1 of 2 NUMBERING PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: Figure 20: Public-Unknown-Numbering Form 15 of 31

16 6. Avaya Modular Messaging In this sample configuration, Avaya Modular Messaging is provisioned for Multi-Site mode. Multi-Site mode allows Avaya Modular Messaging to serve subscribers in multiple locations. The administration for Modular Messaging is beyond the scope of these Application Notes. Refer to [7] and [8] for further details. 7. Configure Acme Packet SBC 1 These Application Notes assume that basic Acme Packet SBC administration has already been performed. In the reference configuration, an Acme Packet 3800 Net-Net Session Director SBC was used. The Acme Packet SBC configuration used in the reference configuration is provided below as a reference. The notable settings are highlighted in bold and brief annotations are provided on the pertinent settings. Consult with Acme Packet Support [9] for further details and explanations on the configuration below. Note - The Sprint SIP Trunk service border element IP addresses shown in this document are examples. Sprint will provide the actual IP addresses as part of the SIP Trunk service provisioning process. 3 Although an Acme Net-Net SD 3800 was used in the reference configuration, theses configurations also apply to the 4250 and 4500 platforms Local Policies ANNOTATION: The local policy below governs the routing of SIP messages from elements on the enterprise network, e.g. Communication Manager, Modular Messaging, etc, to the Sprint SIP Trunk service. The Session Agents are defined here. local-policy from-address * to-address * source-realm Enterprise description activate-time N/A deactivate-time N/A state policy-priority last-modified-by admin@135.x.x.x last-modified-date :18:51 policy-attribute next-hop realm Sprint action terminate-recursion carrier start-time of 31

17 end-time 2400 days-of-week U-S cost 0 app-protocol SIP state methods media-profiles lookup single next-key eloc-str-lkup eloc-str-match local-policy ANNOTATION: The local policy below governs the routing of SIP messages from the Sprint SIP Trunk service to elements on the enterprise network. local-policy from-address * to-address * source-realm Sprint description activate-time N/A deactivate-time N/A state policy-priority last-modified-by admin@135.x.x.x last-modified-date :20:48 policy-attribute next-hop realm Enterprise action terminate-recursion carrier start-time 0000 end-time 2400 days-of-week U-S cost 0 app-protocol SIP state methods media-profiles lookup single next-key eloc-str-lkup eloc-str-match local-policy 7.2. Network Interfaces ANNOTATION: The network interface below defines the IP addresses on the interface connected to the network on which the Sprint SIP Trunk service resides. network-interface name s0p1 sub-port-id 0 description hostname 17 of 31

18 ip-address pri-utility-addr sec-utility-addr netmask gateway sec-gateway gw-heartbeat state heartbeat 0 retry-count 0 retry-timeout 1 health-score 0 dns-ip-primary dns-ip-backup1 dns-ip-backup2 dns-domain dns-timeout 11 hip-ip-list ftp-address icmp-address snmp-address telnet-address ssh-address ANNOTATION: The network interface below defines the IP addresses on the interface connected to the enterprise network on which the Avaya elements reside. network-interface name s0p0 sub-port-id 0 description hostname ip-address pri-utility-addr sec-utility-addr netmask gateway sec-gateway gw-heartbeat state heartbeat 0 retry-count 0 retry-timeout 1 health-score 0 dns-ip-primary dns-ip-backup1 dns-ip-backup2 dns-domain dns-timeout 11 hip-ip-list ftp-address icmp-address snmp-address telnet-address ssh-address 18 of 31

19 7.3. Realms ANNOTATION: The realm configuration Sprint below represents the external network on which the Sprint SIP Trunk service resides. realm-config identifier Sprint description addr-prefix network-interfaces s0p1:0 mm-in-realm mm-in-network mm-same-ip mm-in-system bw-cac-non-mm msm-release generate-udp-checksum max-bandwidth 0 fallback-bandwidth 0 max-priority-bandwidth 0 max-latency 0 max-jitter 0 max-packet-loss 0 observ-window-size 0 parent-realm dns-realm media-policy media-sec-policy in-translationid out-translationid in-manipulationid out-manipulationid manipulation-string manipulation-pattern class-profile average-rate-limit 0 access-control-trust-level invalid-signal-threshold 0 maximum-signal-threshold 0 untrusted-signal-threshold 0 nat-trust-threshold 0 deny-period 30 ext-policy-svr diam-e2-address-realm symmetric-latching pai-strip trunk-context early-media-allow enforcement-profile additional-prefixes restricted-latching restriction-mask 32 accounting-enable user-cac-mode user-cac-bandwidth 0 user-cac-sessions 0 19 of 31

20 icmp-detect-multiplier 0 icmp-advertisement-interval 0 icmp-target-ip monthly-minutes 0 net-management-control delay-media-update refer-call-transfer dyn-refer-term codec-policy codec-manip-in-realm constraint-name call-recording-server-id xnq-state xnq-unknown hairpin-id 0 stun-enable stun-server-ip stun-server-port 3478 stun-changed-ip stun-changed-port 3479 match-media-profiles qos-constraint sip-profile sip-isup-profile block-rtcp hide-egress-media-update ANNOTATION: The realm configuration Enterprise below represents the internal network on which the Avaya elements reside. realm-config identifier Enterprise description addr-prefix network-interfaces s0p0:0 mm-in-realm mm-in-network mm-same-ip mm-in-system bw-cac-non-mm msm-release generate-udp-checksum max-bandwidth 0 fallback-bandwidth 0 max-priority-bandwidth 0 max-latency 0 max-jitter 0 max-packet-loss 0 observ-window-size 0 parent-realm dns-realm media-policy media-sec-policy in-translationid out-translationid in-manipulationid out-manipulationid manipulation-string 20 of 31

21 manipulation-pattern class-profile average-rate-limit 0 access-control-trust-level invalid-signal-threshold 0 maximum-signal-threshold 0 untrusted-signal-threshold 0 nat-trust-threshold 0 deny-period 30 ext-policy-svr diam-e2-address-realm symmetric-latching pai-strip trunk-context early-media-allow enforcement-profile additional-prefixes restricted-latching restriction-mask 32 accounting-enable user-cac-mode user-cac-bandwidth 0 user-cac-sessions 0 icmp-detect-multiplier 0 icmp-advertisement-interval 0 icmp-target-ip monthly-minutes 0 net-management-control delay-media-update refer-call-transfer dyn-refer-term codec-policy codec-manip-in-realm constraint-name call-recording-server-id xnq-state xnq-unknown hairpin-id 0 stun-enable stun-server-ip stun-server-port 3478 stun-changed-ip stun-changed-port 3479 match-media-profiles qos-constraint sip-profile sip-isup-profile block-rtcp hide-egress-media-update 7.4. Session Agents ANNOTATION: The session agent below represents the Sprint SIP Trunk service network border element. The Acme Packet SBC will attempt to send calls to the border element based on successful responses to the OPTIONS ping-method. Redundant network session-agents may be defined. 21 of 31

22 NOTE The ping-method OPTIONS;hops=70 parameter shown below was a setting used in the reference test environment. Acme Packet best practices recommends a setting of OPTIONS;hops=0 in customer deployments. session-agent hostname ip-address port 5060 state app-protocol SIP app-type transport-method UDP realm-id Sprint egress-realm-id description carriers allow-next-hop-lp constraints max-sessions 0 max-inbound-sessions 0 max-outbound-sessions 0 max-burst-rate 0 max-inbound-burst-rate 0 max-outbound-burst-rate 0 max-sustain-rate 0 max-inbound-sustain-rate 0 max-outbound-sustain-rate 0 min-seizures 5 min-asr 0 time-to-resume 0 ttr-no-response 0 in-service-period 0 burst-rate-window 0 sustain-rate-window 0 req-uri-carrier-mode None proxy-mode redirect-action loose-routing send-media-session response-map ping-method OPTIONS;hops=70 ping-interval 60 ping-send-mode keep-alive ping-all-addresses ping-in-service-response-codes out-service-response-codes media-profiles in-translationid out-translationid trust-me request-uri-headers stop-recurse local-response-map ping-to-user-part ping-from-user-part li-trust-me in-manipulationid out-manipulationid manipulation-string 22 of 31

23 manipulation-pattern p-asserted-id trunk-group max-register-sustain-rate 0 early-media-allow invalidate-registrations rfc2833-mode rfc2833-payload 0 codec-policy enforcement-profile refer-call-transfer reuse-connections NONE tcp-keepalive tcp-reconn-interval 0 max-register-burst-rate 0 register-burst-window 0 sip-profile sip-isup-profile ANNOTATION: The session agent below represents the Avaya Aura Communication Manager used in the reference configuration (represented by the IP address of the CLAN card in the Avaya G650 Media Gateway). session-agent hostname ip-address port 5060 state app-protocol SIP app-type transport-method StaticTCP realm-id Enterprise egress-realm-id description carriers allow-next-hop-lp constraints max-sessions 0 max-inbound-sessions 0 max-outbound-sessions 0 max-burst-rate 0 max-inbound-burst-rate 0 max-outbound-burst-rate 0 max-sustain-rate 0 max-inbound-sustain-rate 0 max-outbound-sustain-rate 0 min-seizures 5 min-asr 0 time-to-resume 0 ttr-no-response 0 in-service-period 0 burst-rate-window 0 sustain-rate-window 0 req-uri-carrier-mode None proxy-mode redirect-action Proxy loose-routing send-media-session 23 of 31

24 response-map ping-method OPTIONS ping-interval 60 ping-send-mode keep-alive ping-all-addresses ping-in-service-response-codes out-service-response-codes media-profiles in-translationid out-translationid trust-me request-uri-headers stop-recurse local-response-map ping-to-user-part ping-from-user-part li-trust-me in-manipulationid out-manipulationid manipulation-string manipulation-pattern p-asserted-id trunk-group max-register-sustain-rate 0 early-media-allow invalidate-registrations rfc2833-mode rfc2833-payload 0 codec-policy enforcement-profile refer-call-transfer reuse-connections NONE tcp-keepalive tcp-reconn-interval 0 max-register-burst-rate 0 register-burst-window 0 sip-profile sip-isup-profile 7.5. SIP Configuration ANNOTATION: The sip-config defines global sip-parameters, including SIP timers, SIP options, which realm to send requests to if not specified elsewhere, and enabling the SD to collect statistics on requests other than REGISTERs and INVITEs. sip-config state operation-mode dialog dialog-transparency home-realm-id Enterprise egress-realm-id Enterprise nat-mode None registrar-domain registrar-host registrar-port 0 register-service-route always init-timer of 31

25 max-timer 4000 trans-expire 32 invite-expire 180 inactive-dynamic-conn 32 enforcement-profile pac-method pac-interval 10 pac-strategy PropDist pac-load-weight 1 pac-session-weight 1 pac-route-weight 1 pac-callid-lifetime 600 pac-user-lifetime 3600 red-sip-port 1988 red-max-trans red-sync-start-time 5000 red-sync-comp-time 1000 add-reason-header sip-message-len 4096 enum-sag-match extra-method-stats registration-cache-limit 0 register-use-to-for-lp options max-udp-length=65535 set-inv-exp-at-100-resp refer-src-routing add-ucid-header proxy-sub-events pass-gruu-contact sag-lookup-on-redirect set-disconnect-time-on-bye 7.6. SIP Interfaces ANNOTATION: The SIP interface below is used to communicate with the Sprint SIP Trunk service, and specifies the Sprint realm. Note that the connection between the Acme Packet SBC and the Sprint border element uses UDP. sip-interface state realm-id Sprint description sip-port address port 5060 transport-protocol UDP tls-profile allow-anonymous all ims-aka-profile carriers trans-expire 0 invite-expire 0 max-redirect-contacts 0 proxy-mode redirect-action contact-mode nat-traversal nat-interval of 31

26 tcp-nat-interval 90 registration-caching min-reg-expire 300 registration-interval 3600 route-to-registrar secured-network teluri-scheme uri-fqdn-domain trust-mode all max-nat-interval 3600 nat-int-increment 10 nat-test-increment 30 sip-dynamic-hnt stop-recurse 401,407 port-map-start 0 port-map-end 0 in-manipulationid out-manipulationid manipulation-string manipulation-pattern sip-ims-feature operator-identifier anonymous-priority max-incoming-conns 0 per-src-ip-max-incoming-conns 0 inactive-conn-timeout 0 untrusted-conn-timeout 0 network-id ext-policy-server default-location-string charging-vector-mode pass charging-function-address-mode pass ccf-address ecf-address term-tgrp-mode implicit-service-route rfc2833-payload 101 rfc2833-mode transparent constraint-name response-map local-response-map ims-aka-feature enforcement-profile route-unauthorized-calls tcp-keepalive add-sdp-invite add-sdp-profiles sip-profile sip-isup-profile ANNOTATION: The SIP interface below is used to communicate with the Avaya elements and references the Enterprise realm. Note that TCP is used between the Acme Packet SBC and Avaya Aura Communication Manager. sip-interface state realm-id description Enterprise 26 of 31

27 sip-port address port 5060 transport-protocol TCP tls-profile allow-anonymous all ims-aka-profile carriers trans-expire 0 invite-expire 0 max-redirect-contacts 0 proxy-mode redirect-action contact-mode nat-traversal nat-interval 30 tcp-nat-interval 90 registration-caching min-reg-expire 300 registration-interval 3600 route-to-registrar secured-network teluri-scheme uri-fqdn-domain trust-mode all max-nat-interval 3600 nat-int-increment 10 nat-test-increment 30 sip-dynamic-hnt stop-recurse 401,407 port-map-start 0 port-map-end 0 in-manipulationid out-manipulationid manipulation-string manipulation-pattern sip-ims-feature operator-identifier anonymous-priority max-incoming-conns 0 per-src-ip-max-incoming-conns 0 inactive-conn-timeout 0 untrusted-conn-timeout 0 network-id ext-policy-server default-location-string charging-vector-mode pass charging-function-address-mode pass ccf-address ecf-address term-tgrp-mode implicit-service-route rfc2833-payload 101 rfc2833-mode transparent constraint-name response-map local-response-map ims-aka-feature enforcement-profile 27 of 31

28 route-unauthorized-calls tcp-keepalive add-sdp-invite add-sdp-profiles sip-profile sip-isup-profile 7.7. Steering Pools ANNOTATION: The steering pools below define the IP Addresses and RTP port ranges on the respective realms. The Sprint realm IP Address will be used as the CPE media traffic IP Address to communicate with Sprint (Note: The RTP port range should be obtained from Sprint). Likewise, the IP Address and RTP port range defined for the Enterprise realm steering pool will be used to communicate with the Avaya elements. Please note that the INSIDE realm port range does not have to be within the range specified below. steering-pool ip-address start-port end-port realm-id Sprint network-interface s0p1:0 steering-pool ip-address start-port end-port realm-id Enterprise network-interface s0p0:0 28 of 31

29 8. Verification Steps 8.1. Verification Tests The following steps may be used to verify the configuration: Place an inbound and outbound call, and verify that a two-way talkpath exists, and that the calls remain stable for several minutes and disconnects properly. Place an inbound and outbound call to a phone and verify that the call goes to coverage if it is not answered Troubleshooting Tools The Communication Manager list trace vector, list trace tac, and/or status trunk trunk-groupno or status station extension commands are helpful diagnostic tools to verify correct operation and to troubleshoot problems. MST (Message Sequence Trace) diagnostic traces (performed by Avaya Support) can be helpful in understanding the specific interoperability issues. Various Acme Packet Net-Net SBC troubleshooting tools are available such as, show sipd agents to monitor the status of Session Agents, log files like sipmsg.log to view the details of the SIP messaging, and trace/capture utilities like packet-trace and packet-capture. Consult Acme Packet s documentation for more details on troubleshooting tools, or for information regarding Acme Packet Net-Net Session Director SBCs in general. 9. Conclusion These Application Notes describe the steps for configuring Avaya Aura Communication Manager and Acme Packet Net-Net Session Director SIP trunking with the Sprint SIP Trunk service. The Sprint SIP Trunk service allows enterprises to place outbound calls to the PSTN and to receive calls from the PSTN. The Sprint SIP Trunk Service passed this compliance test. The sample configuration shown in these Application Notes is representative of a basic enterprise customer configuration and is intended to provide configuration guidance to supplement other Avaya product documentation. It is based upon formal interoperability compliance testing as part of the Avaya DevConnect Service Provider program. Please refer to Section 2.3 for any observations or workarounds relating to the testing covered by these Application Notes. 29 of 31

30 10. References Avaya product documentation is available at unless otherwise noted. [1] Administering Avaya Aura Communication Manager, Issue 5.0, Release 5.2, May 2009, Document Number [2] Avaya Aura Communication Manager Feature Description and Implementation, Issue 7, Release 5.2, May 2009, Document Number [3] SIP Support in Avaya Aura Communication Manager Running on the Avaya S8xxx Servers, Issue 9, May 2009, Document Number [4] Avaya one-x Deskphone Edition for 9600 Series SIP IP Telephones Administrator Guide Release 2.5, Issue 5, November 2009, Document Number [5] Avaya Aura Call Center 5.2 Call Vectoring and Expert Agent Selection (EAS) Reference, Release 5.2, April 2009, Document Number [6] Avaya Aura Call Center 5.2 Automatic Call Distribution Reference, Release 5.2, April 2009, Document Number [7] Modular Messaging Multi-Site Guide Release 5.2, November 2009 [8] Modular Messaging for Microsoft Exchange Release 5.2 Installation and Upgrades, Issue 1.0, November 2009 [9] Acme Packet documentation can be found at [10] Sprint SIP Trunk Service information can be found at 30 of 31

31 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 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 31 of 31