Abstract. Avaya Solution & Interoperability Test Lab

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Sotel Systems SIP Trunking with Avaya IP Office Release 8.1 and the Avaya Session Border Controller for Enterprise 6.2 Issue 1.0 Abstract These Application Notes describe the steps to configure Session Initiation Protocol (SIP) Trunking between Sotel Systems Networks and Avaya IP Office Release 8.1 with the Avaya Session Border Controller for Enterprise 6.2. Sotel Systems SIP Trunking provides PSTN access via a SIP trunk between the enterprise and the Sotel Systems network as an alternative to legacy analog or digital trunks. This approach generally results in lower cost for the enterprise. Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted in the Avaya Solutions and Interoperability Test Lab, utilizing Sotel Systems SIP Trunk Services. 1 of 69

2 1. Introduction These Application Notes describe the steps required to configure Session Initiation Protocol (SIP) Trunking between Sotel Systems and Avaya IP Office Release 8.1 with the Avaya Session Border Controller for Enterprise 6.2. The Sotel Systems SIP Trunking service referenced within these Application Notes is positioned for customers that have an IP-PBX or IP-based network equipment with SIP functionality, but need a form of IP transport and local services to complete their solution. Sotel Systems SIP Trunking will enable delivery of origination and termination of local, longdistance and toll-free traffic across a single broadband connection. A SIP signaling interface will be enabled to the Customer Premises Equipment (CPE). 2. General Test Approach and Test Results The general test approach was to connect a simulated enterprise site to the Sotel Systems 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 IP Office and various Avaya endpoints. 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 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 including H.323, SIP, digital, and analog telephones at the enterprise. All inbound PSTN calls were routed to the enterprise across the SIP trunk from the service provider Outgoing PSTN calls to various phone types including H.323, SIP, digital, and analog telephones at the enterprise. All outgoing PSTN calls were routed to the enterprise across the SIP trunk from the service provider Inbound and outbound PSTN calls to/from Avaya IP Office Softphone Various call types including: local, long distance, outbound toll-free, and local directory assistance Codecs G.729A and G.711MU 2 of 69

3 T.38 Fax Caller ID presentation and Caller ID restriction DTMF transmission using RFC 2833 Voic navigation using DTMF for inbound and outbound calls User features such as hold and resume, transfer, and conference Off-net call forwarding and twinning 2.2. Test Results Interoperability testing of Sotel Systems SIP Trunking was completed with successful results for all test cases with the exception of the observations/limitations described below. International dialing, Information via 411 and 1411, Operator and Operator Assisted calls were not supported in the Sotel Systems lab environment and therefore were not tested as part of this compliance test. When an inbound or outbound call was left to ring with no answer, at about three minutes the caller would get dead air briefly and then a fast-busy. This isn t proper but was also considered a minor issue Support For technical support on Sotel Systems SIP Trunking, contact Sotel Systems at 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. 3 of 69

4 3. Reference Configuration Figure 1 illustrates the sample configuration used for the DevConnect compliance testing. The sample configuration shows an enterprise site connected to Sotel Systems SIP Trunking. Located at the enterprise site is an Avaya IP Office 500 V2. The LAN port of Avaya IP Office is connected to the enterprise LAN while the WAN port is connected to the private side of the Avaya Session Border Controller for Enterprise (SBCE). The Avaya SBCE in turn is connected to Sotel Systems SIP trunk service via its WAN port. Endpoints include an Avaya 1616 IP Telephone (with H.323 firmware), an Avaya 1140E IP Telephone (with SIP firmware), the Avaya IP Office Softphone, an Avaya 1408 Digital Telephone, an Avaya Analog Telephone and the Ventafax application running on a PC. The site also has a Windows 2003 Server running Avaya Voic Pro for voic and the Avaya IP Office Manager to configure the Avaya IP Office. Figure 1: Avaya Interoperability Test Lab Configuration 4 of 69

5 For security purposes, any public IP addresses or PSTN routable phone numbers used in the compliance test are not shown in these Application Notes. Instead, public IP addresses have been replaced with private addresses and all phone numbers have been replaced with numbers that cannot be routed. In an actual customer configuration, the enterprise site may also include additional network components between the service provider and the Avaya IP Office such as a data firewall. A complete discussion of the configuration of these types of devices is beyond the scope of these Application Notes. However, it should be noted that SIP and RTP traffic between the service provider and the Avaya SBCE, and then between the Avaya SBCE and IP Office, must be allowed to pass through these devices. 4. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Avaya IP Telephony Solution Components Equipment Software Avaya IP Office 500 V2 Release 8.1 (63) Avaya Voic Pro Release 8.1 (810) Avaya IP Office Manager Release 10.1 (63) Avaya 1616 IP Telephone (H.323) Release 1.301S Avaya 1140E IP Telephone (SIP) Release Avaya 1408 Digital Telephone Release 0.39 Avaya Analog Telephone N/A Avaya IP Office Softphone Release (67009) Avaya Session Border Controller for Release 6.2 Enterprise Sotel Systems SIP Trunking Solution Components Component Release Sotel Systems CCNQ Configure Avaya IP Office Avaya IP Office is configured through the Avaya IP Office Manager PC application. From the Avaya IP Office Manager PC, select Start Programs IP Office Manager to launch the application. Navigate to File Open Configuration, select the proper Avaya IP Office system from the pop-up window and log in with the appropriate credentials. The appearance of the IP Office Manager can be customized using the View menu. In the screens presented in this section, the View menu was configured to show the Navigation pane on the left side, the Group pane in the center, and the Details pane on the right side. These panes will be referenced throughout the Avaya IP Office configuration. All licensing and feature configuration that is not directly related 5 of 69

6 to the interface with the service provider (such as twinning and IP Office Softphone support) is assumed to already be in place Licensing and Physical Hardware The configuration and features described in these Application Notes require the IP Office system to be licensed appropriately. If a desired feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales representative. To verify that there is a SIP Trunk Channels License with sufficient capacity; click License in the Navigation pane and SIP Trunk Channels in the Group pane. Confirm there is a valid license with sufficient Instances (trunk channels) in the Details pane. 6 of 69

7 If Avaya IP Telephones will be used, as is the case in these Application Notes, verify the Avaya IP endpoints license. Click License in the Navigation pane and Avaya IP endpoints in the Group pane. Confirm there is a valid license with sufficient Instances in the Details pane. 7 of 69

8 In the sample configuration, looking at the IP Office 500v2 from left to right, the first module is a Combination Card. This module has 6 Digital Stations ports, 2 Analog Station ports, 4 Analog Trunk ports and 10 VCM channels. The VCM32 module is a Voice Compression Module supporting VoIP codecs. An IP Office hardware configuration with a VCM component is necessary to support SIP trunking. The third module is a TCM 8. This module supports BCM / Norstar T-Series and M-Series telephones. The following screen shows the modules populated in the IP Office system used in the sample configuration. To access such a screen, select Control Unit in the Navigation pane. The modules appear in the Group pane. In the screen below, IP 500 V2 is selected in the Group pane, revealing additional information about the IP 500 V2 in the Details pane. 8 of 69

9 5.2. LAN1 Settings In the sample configuration the WAN port was used to connect the Avaya IP Office to the public network. The LAN1 settings correspond to the LAN port on the back of the Avaya IP Office 500v2. To access the LAN1 settings, first navigate to System in the Navigation Pane and then navigate to the LAN1 LAN Settings tab in the Details Pane. Set the IP Address field to the IP address assigned to the Avaya IP Office LAN port. Set the IP Mask field to the mask used on the private network. All other parameters should be set according to customer requirements. 9 of 69

10 On the VoIP tab in the Details Pane, check the SIP Trunks Enable box to enable the configuration of SIP trunks and then check the H.323 Gatekeeper Enable and SIP Registrar Enable boxes to allow H.323 and SIP handsets to register to the system. The RTP Port Number Range can be customized to a specific range of receive ports for the RTP media. Based on this setting, Avaya IP Office would request RTP media be sent to a UDP port in the configurable range for calls using LAN1. Avaya IP Office can also be configured to mark the Differentiated Services Code Point (DSCP) in the IP Header with specific values to support Quality of Services policies for both signaling and media. The DSCP field is the value used for media and the SIG DSCP is the value used for signaling. The specific values used for the compliance test are shown in the example below and are also the default values. All other parameters should be set according to customer requirements. Scroll to the bottom of the Details Pane. Under RTP Keepalives set the Scope to RTP, the Initial keepalives to Enabled and the Periodic timeout to 10. Enabling this will prevent the loss of speech path on calls forwarded across the SIP trunk. These settings instruct Avaya IP Office to send RTP keepalive packets every 10 seconds from the establishment of the connection. This will start media flowing from the far-end endpoint in those cases where the far-end endpoint is waiting to receive media before it starts to send media of its own. 10 of 69

11 11 of 69

12 On the Network Topology tab in the Details Pane, configure the following parameters: Select the Firewall/NAT Type from the pull-down menu that matches the network configuration. No firewall or network address translation (NAT) device was used in the compliance test as shown in Figure 1, so the parameter was set to Open Internet. Set Binding Refresh Time (seconds) to 60. This value is used to determine the frequency at which Avaya IP Office will send SIP OPTIONS messages to the service provider. Set Public IP Address to the IP address of the Avaya IP Office LAN port. Set the Public Port to All other parameters should be set according to customer requirements. 12 of 69

13 5.3. Voic Settings On the Voic tab in the Details Pane, configure the SIP Settings section. The values entered for the SIP Name and Contact fields are used as the user part of the SIP URI in the From and Contact headers for outgoing SIP trunk calls from Voic (e.g., Outcalling). The SIP Name and Contact are set to one of the DID numbers assigned to the enterprise from Sotel Systems. The SIP Display Name (Alias) parameter can optionally be configured with a descriptive name (e.g. Voic ). Uncheck the Anonymous box to allow Voic Caller ID information to the network. 13 of 69

14 5.4. System Telephony Settings On the Telephony tab in the Details Pane, uncheck the Inhibit Off-Switch Forward/Transfer box to allow call forwarding and call transfers to the PSTN via the service provider across the SIP trunk. If for security reasons incoming calls should not be allowed to transfer back to the PSTN then leave this setting checked Twinning Calling Party Settings To view or change Twinning settings, select the Twinning tab as shown in the following screen. The Send original calling party information for Mobile Twinning box is not checked in the sample configuration, and the Calling party information for Mobile Twinning is left blank. This will allow the Send Caller ID setting in Section to control the calling party information for Mobile Twinning and forwarded calls. 14 of 69

15 5.6. IP Route Though not explicitly necessary in this configuration, a default route can be set for the LAN1 interface. Navigate to IP Route in the left Navigation Pane, and then right-click on the Group Pane to select New (not shown). Create a default route with the following parameters: Set IP Address and IP Mask to Set Gateway IP Address to the IP Address of the default router to reach the Avaya SBCE. Set Destination to LAN1 from the pull-down menu. 15 of 69

16 5.7. SIP Line A SIP line is needed to establish the SIP connection between Avaya IP Office and Sotel Systems SIP Trunking. To create a SIP line, begin by navigating to Line in the Navigation Pane. Rightclick in the Group Pane and select New SIP Line (not shown) SIP Line SIP Line Tab On the SIP Line tab in the Details Pane, configure the parameters as shown below. Leave ITSP Domain Name field blank. Set Send Caller ID to Diversion Header. With this setting and the related configuration in Section 5.5, IP Office will include the Diversion Header for calls that are directed via Mobile Twinning out the SIP Line to Sotel Systems. It will also include the Diversion Header for calls that are call forwarded out the SIP Line. Under REFER Support, set the Incoming and Outgoing fields to Never and uncheck Refer Support. See Section 2.2 regarding limitations observed using SIP REFER. Check the In Service box. This makes the trunk available to incoming and outgoing calls. Check the Check OOS box. IP Office will use the SIP OPTIONS method to periodically check the SIP Line. The time between SIP OPTIONS sent by IP Office will use the Binding Refresh Time for LAN1 as shown in Section 5.2. Default values may be used for all other parameters. 16 of 69

17 SIP Line - Transport Tab Select the Transport tab. This tab was first introduced in Release 6.1. Some information configured in this tab had been under the SIP Line tab in Release 6.0. Set the parameters as shown below. Set ITSP Proxy Address to the IP address of the private interface of the Avaya SBCE. In the compliance test, outbound calls were placed by setting up the 9N short code without an IP address in the Telephone Number field, as detailed in Section 5.8. Set Layer 4 Protocol to TCP Set Use Network Topology Info to the network port configured in Section 5.2. Set the Send Port to Default values may be used for all other parameters. 17 of 69

18 SIP Line - SIP URI Tab A SIP URI entry must be created to match each incoming number that Avaya IP Office will accept on this line. Select the SIP URI tab, then click the Add button and the New Channel area will appear at the bottom of the pane. To edit an existing entry, click an entry in the list at the top, and click the Edit button. In the example screen below, a previously configured entry is edited. The entry was created with the parameters shown below: Set Local URI, Contact, Display Name, and PAI to Use Internal Data. This setting allows calls on this line when the SIP URI matches the number set in the SIP tab of any User as shown in Section 5.9. Associate this line with an incoming line group by entering a line group number in the Incoming Group field. This line group number will be used in defining incoming call routes for this line. Similarly, associate the line to an outgoing line group using the Outgoing Group field. The outgoing line group number is used in defining short codes for routing outbound traffic to this line. For the compliance test, a new incoming and outgoing group 18 was defined that only contains this line (Line 18). Set Max Calls per Channel to the number of simultaneous SIP calls that are allowed using this SIP URI pattern. 18 of 69

19 In the sample configuration, the single SIP URI previously created, which is shown below as Channel 1, was sufficient to allow incoming calls for Sotel Systems DID numbers destined for specific IP Office users or IP Office hunt groups. These calls are accepted by IP Office since the incoming number will match the SIP Name configured for the user or hunt group that is the destination for the call. Channel 2 displays this service number, which could be used as a DID number that is routed directly to voic or used for Mobile Call Control such as FNE31. DID numbers that IP Office should admit can be entered into the Local URI and Contact fields instead of Use Internal Data. The number will be assigned as a service number in the Incoming Call Routes in Section Note that only the Incoming Line Group was populated for these service numbers as these numbers are only used for inbound call services. 19 of 69

20 SIP Line - VoIP Tab Select the VoIP tab to set the Voice over Internet Protocol parameters of the SIP line. Set the parameters as shown below. Set the Codec Selection field to Custom to allow the specific codec selection to be different from the system default. To modify, click on a codec and use the arrow keys to move it to the Selected column and to change the order of preference. For the compliance test, G.729A and G.711MU were used. Uncheck the VoIP Silence Suppression box. Check the Re-invite Supported box. Set the Fax Transport Support to T.38. Set the DTMF Support field to RFC2833. This directs Avaya IP Office to send DTMF tones using RTP events messages as defined in RFC2833. Default values may be used for all other parameters. Click the OK button at the bottom of the page (not shown). 20 of 69

21 5.8. Short Codes Define a short code to route outbound traffic to the SIP line. To create a short code, right-click on Short Code in the Navigation Pane and select New (not shown). On the Short Code tab in the Details Pane, configure the parameters as shown below. In the Code field, enter the dial string which will trigger this short code, followed by a semi-colon. In this case, 9N;. This short code will be invoked when the user dials 9 followed by any number. Set Feature to Dial. This is the action that the short code will perform. Set Telephone Number to N. This field is used to construct the Request URI and To headers in the outgoing SIP INVITE message. The value N represents the number dialed by the user. Set the Line Group Id to the outgoing line group number defined on the SIP URI tab on the SIP Line in Section This short code will use this line group when placing the outbound call. Click the OK button (not shown). 21 of 69

22 The simple 9N; short code previously illustrated does not provide a means of alternate routing if the configured Sotel Systems SIP Line is out of service or temporarily not responding. When alternate routing options and/or more customized analysis of the digits following the short code are desired, the Automatic Route Selection (ARS) feature may be used. In the following example screen, the short code 8N is illustrated for access to ARS. When the Avaya IP Office user dials 8 plus any number N, rather than being directed to a specific Line Group Id, the call is directed to 50: Main, configurable via ARS. See Section 5.11 for example ARS route configuration for 50: Main as well as a backup route. Optionally, add or edit a short code that can be used to access the SIP Line anonymously. In the screen shown below, the short code *67N; is illustrated. This short code is similar to the 9N; short code except that the Telephone Number field begins with the letter W, which means withhold the outgoing calling line identification. In the case of the SIP Line to Sotel Systems documented in these Application Notes, when a user dials *67 plus any number N, IP Office will include the user s telephone number in the P-Asserted-Identity (PAI) header along with Privacy: Id. Sotel Systems will allow the call due to the presence of a valid DID in the PAI header, but will prevent presentation of the caller id to the called PSTN destination. 22 of 69

23 The following screen illustrates a short code that acts like a feature access code rather than a means to access a SIP Line. In this case, the Code FNE31 is defined for Feature FNE Service to Telephone Number 31 (Mobile Call Control). This short code will be used as means to allow a Sotel Systems DID to be programmed to route directly to this feature, via inclusion of this short code as the destination of an Incoming Call Route. See Section This feature is used to provide dial tone to twinned mobile devices (e.g., cell phone) directly from IP Office; once dial tone is received the user can perform dialing actions including making calls and activating Short Codes. 23 of 69

24 5.9. User Configure the SIP parameters for each user that will be placing and receiving calls via the SIP line defined in Section 5.7. To configure these settings, first navigate to User in the Navigation Pane, and then click on the user in the Group Pane to be modified. Select the SIP tab in the Details Pane. The values entered for the SIP Name and Contact fields are used as the user part of the SIP URI in the From and Contact headers for outgoing SIP trunk calls and allow matching of the SIP URI for incoming calls without having to enter this number as an explicit SIP URI for the SIP line. See Section The example below shows the settings for User 301. The SIP Name and Contact are set to one of the DID numbers assigned to the enterprise from Sotel Systems. The SIP Display Name (Alias) parameter can optionally be configured with a descriptive name. If all calls involving this user and a SIP Line should be considered private, then the Anonymous box may be checked to withhold the user s information from the network. Click the OK button (not shown). 24 of 69

25 The following screen shows the Mobility tab for User 301. The Mobility Features and Mobile Twinning boxes are checked. The Twinned Mobile Number field is configured with the number to dial to reach the twinned mobile telephone over the SIP trunk, in this case Other options can be set according to customer requirements. 25 of 69

26 5.10. Incoming Call Route An incoming call route maps an inbound DID number on a specific line to an internal extension. This procedure can be repeated for each DID number provided by the service provider or the procedure that follows can be used for simplicity. To create an incoming call route, right-click Incoming Call Routes in the Navigation Pane and select New. On the Standard tab of the Details Pane, enter the parameters as shown below. Set the Bearer Capacity to Any Voice. Set the Line Group Id to the incoming line group of the SIP line defined in Section Leave the Incoming Number field blank. This, in conjunction with the configuration of the Destination tab shown in the step below, will look to match an incoming number on the respective Line Group ID with the DIDs populated on the User SIP tab as shown in Section 5.9 above. Default values can be used for all other fields. On the Destinations tab, set the Destination field to. by typing a period in the field (do not use the pull down menu). This will cause the system to look for a match on the incoming DID within the User SIP tab(s) configured in Section 5.9. Click the OK button (not shown). 26 of 69

27 Incoming Call Routes for other direct mappings of DID numbers to IP Office users listed in Figure 1 are omitted here, but can be configured in the same fashion. In the screen shown below, the incoming call route for Incoming Number is illustrated. The Line Group Id is 18, matching the Incoming Group field configured in the SIP URI tab in Section When configuring an Incoming Call Route, the Destination field can be manually configured with a number such as a short code, or certain keywords available from the pull-down menu. For example, the following Destinations tab for an incoming call route contains the Destination FNE31 entered manually. FNE31 is the short code for FNE Service, as shown in Section 5.8. An incoming call to will be delivered directly to internal dial tone from the IP Office, allowing the caller to perform dialing actions including making calls and activating Short Codes. The incoming caller ID must match the Twinned Mobile Number entered in the User Mobility tab (Section 5.9); otherwise the IP Office responds with a 486 Busy Here and the caller will hear a busy tone. An Incoming Call Route can be similarly configured to access the voic system using a Short Code with a Feature of Voic Collect as the Destination. 27 of 69

28 5.11. ARS and Alternate Routing While detailed coverage of ARS is beyond the scope of these Application Notes, this section includes basic ARS screen illustrations and considerations. ARS is shown here mainly to give an example of alternate routing should the SIP Line be out of service or temporarily not responding. Optionally, Automatic Route Selection (ARS) can be used rather than the simple 9N; short code approach documented in Section 5.8. With ARS, secondary dial tone can be provided after the access code, time-based routing criteria can be introduced, and alternate routing can be specified so that a call can re-route automatically if the primary route or outgoing line group is not available. ARS also facilitates more specific dialed telephone number matching, enabling immediate routing and alternate treatment for different types of numbers following the access code. For example, if all local and long distance calls should use the SIP Line preferentially, but service numbers should prefer a different outgoing line group, ARS can be used to distinguish the call behaviors. To add a new ARS route, right-click ARS in the Navigation pane and select New. To view or edit an existing ARS route, select ARS in the Navigation pane and select the appropriate route name in the Group pane (not shown). The following screen shows an example ARS configuration for the route named Main. The In Service parameter refers to the ARS form itself, not the Line Groups that may be referenced in the form. If the In Service box is un-checked, calls are routed to the ARS route name specified in the Out of Service Route parameter. IP Office short codes may also be defined to allow an ARS route to be disabled or enabled from a telephone. The configurable provisioning of an Out of Service Route and the means to manually activate the Out of Service Route can be helpful for scheduled maintenance or other known service-affecting events for the primary route. 28 of 69

29 Assuming the primary route is in-service, the number passed from the short code used to access ARS (e.g., 8N in Section 5.8) can be further analyzed to direct the call to a specific Line Group ID per the example screen above. If the user dialed , the call would be directed to Line Group 18, configured in Section If Line Group 18 cannot be used, the call can automatically route to the route name configured in the Alternate Route parameter in the lower right of the screen. Since alternate routing can be considered a privilege not available to all callers, IP Office can control access to the alternate route by comparing the calling user s priority to the value in the Alternate Route Priority Level field. 29 of 69

30 The following screen shows an example ARS configuration for the route named backup, ARS Route ID 51. Continuing the example, if the user dialed , and the call could not be routed via the primary route 50: Main described above, the call will be delivered to this Alternate Route. Per the configuration shown below, the call will be delivered to Line Group ID 1, using the analog lines. The configuration of the Code, Telephone Number, Feature, and Line Group ID for an ARS route is similar to the configuration already shown for short codes in Section of 69

31 5.12. Privacy/Anonymous Calls For outbound calls with privacy (anonymous) enabled, Avaya IP Office will replace the calling party number in the From and Contact headers of the SIP INVITE message with restricted and anonymous respectively. Avaya IP Office can be configured to use the P-Preferred-Identity (PPI) or P-Asserted-Identity (PAI) header to pass the actual calling party information for authentication and billing. By default, Avaya IP Office will use PPI for privacy. For the compliance test, PAI was used for the purposes of privacy. To configure Avaya IP Office to use PAI for privacy calls, navigate to User in the Navigation Pane, then NoUser in the Group Pane. Select the Source Numbers tab in the Details Pane. Click the Add button. At the bottom of the Details Pane, the Source Number field will appear. Enter SIP_USE_PAI_FOR_PRIVACY. Click OK. The SIP_USE_PAI_FOR_PRIVACY parameter will appear in the list of Source Numbers as shown below. Click OK at the bottom of the screen (not shown). 31 of 69

32 5.13. Save Configuration Navigate to File Save Configuration in the menu bar at the top of the screen to save the configuration performed in the preceding sections. The following will appear, with either Merge or Immediate selected, based on the nature of the configuration changes made since the last save. Note that clicking OK may cause a service disruption. Click OK if desired. 32 of 69

33 6. Configure Avaya Session Border Controller for Enterprise In the sample configuration, an Avaya SBCE is used as the edge device between the Avaya CPE and the Sotel Systems SIP Trunking service. These Application Notes assume that the installation of the Avaya SBCE and the assignment of a management IP Address have already been completed Access Management Interface Use a WEB browser to access the web management interface by entering URL where <ip-address> is the management LAN IP address assigned during installation. Log in using proper login credentials (not shown). Once logged in, a welcome screen will be presented. The following image illustrates the menu items available on the left-side of the UC-Sec Dashboard screen. 33 of 69

34 6.2. System Status Navigate to Dashboard System Management. A list of installed devices is shown in the right pane. For the sample configuration, a single device named ASBCE-1 is shown. Device Status Commissioned should be displayed as shown below. To view the network information of this device, which was assigned during installation, click the View option (the third entry from the right). A System Information window is displayed as shown below. Note that the A1 and B1 interface IP addresses correspond to the inside and outside interfaces, respectively, for the Avaya SBCE as shown in Figure of 69

35 6.3. DNS Settings Since Sotel Systems uses a fully qualified domain name which returns two different A records, it is necessary to setup Domain Name Service (DNS) on the Avaya SBCE so that calls are accepted from both SIP proxy IP addresses. To setup DNS navigate to System ManagementDevices and click on the Edit button. The following screen is displayed. 35 of 69

36 In this example, the Primary DNS has been set to the local area network s DNS server, and the DNS Client IP has been set to the Avaya SBCE s LAN interface, interface A1. A Secondary DNS server can be populated as well if desired. NOTE: In a customer environment, it may be necessary to change the DNS Client IP field to another interface depending on where the DNS server is located Global Profiles Server Interworking Server Interworking is defined for each server connected to the Avaya SBCE. For the compliance test, the Sotel Systems SIP trunk network-edge SBC serves as the Trunk Server and IP Office serves as the Call Server. Navigate to Global Profiles Server Interworking from the left-side menu (not shown) to configure Server Interworking profiles Server Interworking: IP Office Click the Add Profile button (not shown) to add a new profile or select an existing Server Interworking profile to edit. If adding a profile, a screen such as the following is displayed. Enter an appropriate Profile Name such as IPO-Interwrk shown below. Click Next. The following screens illustrate the General parameters used in the sample configuration for the Interworking Profile named IPO-Interwrk. Most parameters retain default values. In the sample configuration, T.38 Support was unchecked as Sotel Systems does not support T.38 faxing, and Hold Support was set for RFC of 69

37 37 of 69

38 Click Next to advance to configure Privacy and DTMF general parameters, which can retain default values. The parameters in all other tabs may retain default settings. Click Finish when done, or Back to edit a previous entry Server Interworking: ServiceProvider A second Server Interworking profile named ServiceProvider was similarly created. Click the Add Profile button (not shown) to add a new profile or select an existing Server Interworking profile to edit. If adding a profile, a screen such as the following is displayed. Enter an appropriate Profile Name such as ServiceProvider as shown below. Click Next. The following screen illustrates the General parameters used in the sample configuration for the ServiceProvider Server Interworking profile. In the sample configuration, T.38 Support was unchecked and Hold Support was set for RFC3264. Other parameters can retain their default values. 38 of 69

39 Click Next to advance to configure Privacy and DTMF general parameters, which can retain default values. The parameters in all other tabs may retain default settings. Click Finish when done, or Back to edit a previous entry. 39 of 69

40 6.5. Global Profiles Server Configuration In the compliance test, the Sotel Systems SIP trunk network-edge SBC is connected as the Trunk Server and IP Office is connected as the Call Server. Navigate to Global Profiles Server Configuration from the left-side menu to configure the two servers Server Configuration for IP Office Click the Add Profile 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 IP Office shown below. Click Next. The following screens illustrate the Server Configuration with Profile name IP Office. Select Call Server from the Server Type drop-down menu. In the IP Addresses / Supported FQDNs area, the IP Address of the IP Office SIP signaling interface should be entered. In the Supported Transports area, TCP is selected and the TCP Port is set to If adding a new profile, click Next (not shown). If editing an existing profile, click Finish. 40 of 69

41 Once configuration is completed, the General tab for the configured IP Office call server will appear as shown below: If adding the profile, click Next to accept default parameters for the Authentication tab, and advance to the Heartbeat area (not shown). If editing an existing profile, select the Heartbeat tab and click Edit. The Avaya SBCE can be configured to source heartbeats in the form of SIP OPTIONS. In the sample configuration, with one connected IP Office, this configuration is optional. If SBC-sourced OPTIONS messages are desired, check the Enable Heartbeat box. Select OPTIONS from the Method drop-down menu. Select the desired Frequency (in seconds) that the SBC 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 SBC toward IP Office. If adding a new profile, click Next (not shown). If editing an existing profile, click Finish. 41 of 69

42 If SBC sourced OPTIONS are configured, the Heartbeat tab for the IP Office server profile will appear as shown below: If adding a profile, click Next to continue to Advanced settings. If editing an existing profile, select the Advanced tab and click Edit. In the resultant screen, select the Interworking Profile IPO-Interwrk created in Section Click Finish. Once configuration is completed, the Advanced tab for the call server IP Office will appear as shown below. 42 of 69

43 Server Configuration for Sotel Systems SIP Trunking Two Server Configuration profiles named SP-SIPT and SP-SIPT2 were similarly created. It was necessary to create two Server Configurations for the Service Provider since Sotel Systems returns two SIP proxy IP addresses for the FQDN termination2.sotelips.net. The following screens illustrate the SP-SIPT and SP-SIPT2 Server Configuration profiles. In the General parameters, select Trunk Server from the Server Type drop-down menu. In the IP Addresses / Supported FQDNs area, enter the IP address for the respective Sotel Systems SIP proxy. In the Supported Transports area, UDP is selected and the UDP Port is set to 5060 as specified by Sotel Systems. Server Configuration SP-SIPT is shown below: Below is the Server Configuration SP-SIPT2: 43 of 69

44 If adding the profile, click Next to accept default parameters for the Authentication tab, and advance to the Heartbeat area (not shown). If editing an existing profile, select the Heartbeat tab and click edit. The Avaya SBCE can be configured to source heartbeats in the form of SIP OPTIONS towards Sotel Systems. This configuration is optional. Independent of whether the SBC is configured to source SIP OPTIONS towards Sotel Systems, Sotel Systems will receive OPTIONS from IP Office. When IP Office sends SIP OPTIONS to the inside private IP Address of the Avaya SBCE, the Avaya SBCE will forward those SIP OPTIONS to Sotel Systems. When Sotel Systems responds, the SBC will pass the response back to IP Office. If SBC-sourced OPTIONS are desired, select OPTIONS from the Method drop-down menu. Select the desired Frequency that the SBC 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 SBC. If adding a new profile, click Next (not shown). If editing an existing profile, click Finish. An example configuration for sourcing SIP OPTIONS from the Avaya SBCE towards the service provider, is shown below. If the optional SBC sourced OPTIONS configuration is completed, the Heartbeat tab for the SP-SIPT server profile will appear as shown below. 44 of 69

45 If adding a profile, click Next to continue to Advanced settings (not shown). If editing an existing profile, select the Advanced tab and click Edit. In the resultant screen, select the Interworking Profile ServiceProvider created in Section Click Finish. Once configuration is completed, the Advanced tab for SP-SIPT and SP-SIPT2 will appear as shown below Global Profiles Routing Routing information is required for traffic to be routed to IP Office on the internal side, and to the Sotel Systems network on the external side. The IP addresses and ports defined here will be used as the destination addresses for signaling. If no port is specified, the default SIP port of 5060 is used. Navigate to Global Profiles Routing from the left-side menu to configure Routing profiles. 45 of 69

46 Routing Configuration for IP Office Click the Add Profile button (not shown) to add a new profile, or select an existing routing profile to edit. If adding a profile, a screen such as the following is displayed. Enter a Profile Name such as To_Avaya shown below. Click Next. In the Next Hop Routing configuration, enter the IP Address of the IP Office SIP signaling interface with port number (optional if port number is 5060) as Next Hop Server 1, as shown below. Check Routing Priority based on Next Hop Server. Choose TCP for Outgoing Transport. 46 of 69

47 Once configuration is completed, the Routing Profile for To_Avaya will appear as follows: Routing Configuration for Sotel Systems SIP Trunking A Routing Profile named To_ServiceProvider for the trunk server was similarly configured as shown below. Click the Add Profile button (not shown) to add a new profile, or select an existing routing profile to edit. If adding a profile, a screen such as the following is displayed. Enter a Profile Name such as To_ServiceProvider shown below. Click Next. In the Next Hop Routing configuration, enter the IP Address of the Sotel Systems SIP trunking SBC signaling interface with port number (again, optional if port number is 5060) as Next Hop Server 1, as shown below. Check Routing Priority based on Next Hop Server. Choose UDP for Outgoing Transport as Sotel Systems accepts SIP traffic over UDP. 47 of 69

48 Once configuration is completed, the Routing Profile for To_ServiceProvider will appear as follows: 6.7. Global Profiles Topology Hiding Topology Hiding is a security feature which allows the changing of several parameters within SIP packets, preventing the private enterprise network information from being propagated to the un-trusted public network. Topology Hiding can also be used as an interoperability tool to adapt certain parameters in selected SIP headers to meet expectations by IP Office and the SIP trunk service provider, allowing the call to be accepted in each case. For the compliance test, only the minimum configuration required to achieve interoperability was performed. 48 of 69

49 Navigate to Global Profiles Topology Hiding from the left-side menu for configuring Topology Hiding profiles Topology Hiding for IP Office Click the Add Profile 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 IP-Office shown below. Click Next. In the resultant screen, click the Add Header button to reveal additional headers. To ensure that the domain received by IP Office from the SBC is the expected enterprise domain, select Overwrite from the drop-down menu as the Replace Action for the To and Request-Line headers. Enter the enterprise domain in the Overwrite Value column as shown below. In the example below, the domain received by IP Office has been changed by the Avaya SBCE to avayalab2.com. Click Finish. 49 of 69

50 After configuration is completed, the Topology Hiding for profile IP-Office will appear as follows. 50 of 69

51 Topology Hiding for Sotel Systems SIP Trunking A Topology Hiding profile named ServiceProvider for Sotel Systems was similarly configured as shown below. Note that it was not necessary to configure any Topology Hiding for SIP signaling that was forwarded to the Service Provider. Default values were used for all fields Domain Policies Media Rules Domain Policies allow configuring, managing and applying various sets of rules designed to control and normalize the behavior of call flows based upon various criteria of communication sessions originating from or terminating to the enterprise. Navigate to Domain Policies Media Rules from the left-side menu to configure Media Rules. In the sample configuration, a single media rule was used. This media rule was cloned from the default rule default-low-med by selecting the default rule default-low-med then clicking the Clone Rule button in the upper right corner as shown below: Enter a descriptive Clone Name and then click Finish. 51 of 69

52 The cloned media rule will be displayed in the Media Rules list on the left. Select this cloned rule from the list, then select the Media Anomaly tab. In the displayed Media Anomaly tab, verify that Media Anomaly Detection is unchecked as shown below. If not, click Edit and uncheck Media Anomaly Detection. The rule named modified-dft-low-med was used during the compliance test Domain Policies End Point Policy Groups End Point Policy Groups are associations of different sets of rules (Media, Signaling, Security, etc ) to be applied to specific SIP messages traversing the Avaya SBCE. Navigate to Domain Policies End Point Policy Groups from the left-side menu to configure End Point Policy Groups. Select the Add button above the list of Policy Groups. Enter a name in the Group Name field, such as Avaya as shown below. Click Next. 52 of 69

53 In the sample configuration, defaults were selected for all fields, with the exception of: Media Rule, which was set to the modified-dft-low-med media rule as defined in Section 7.7 Signaling Rule, which was set to the Avaya_SigRules signaling rule as defined in Section 7.8 Click Finish. Once configuration is completed, the Avaya End Point Policy Group will appear as follows. 53 of 69

54 Repeat the configuration steps above to create a second End Point Policy Group named ServiceProvider for the network side as shown below. Note that this End Point Policy Group uses the same Media Rule (modified-dft-low-med) for disabling Media Anomaly Detection and the default Signaling Rule since no header manipulations are required for messages to and from the outside interface of the Avaya SBCE Device Specific Settings Network Management The network information should have been previously specified during the installation of the Avaya SBCE. Navigate to Device Specific Setting Network Management from the left-side menu. Under Devices, select the device being managed, which was named ASBCE-1 in the sample configuration. The Network Configuration tab is shown below. Observe the IP Address, Netmask, Gateway, and Interface information previously assigned. Note that only the A1 and 54 of 69

55 B1 interfaces are used, typically the A interfaces are used for the internal side and B interfaces are used for the external side of the Avaya SBCE. Select the Interface Configuration tab. The Administrative Status can be toggled between Enabled and Disabled in this screen. The following screen was captured after the interfaces had already been enabled. To enable the interface if it is disabled, click the Toggle State button. When IP addresses and network masks are assigned to interfaces, these are then configured as signaling and media interfaces Device Specific Settings Media Interface Media Interfaces are created to adjust the port range assigned to media streams leaving the interfaces of the SBC. The compliance test used the port range to for the inside, private interface which is the default media port range for Avaya IP Office. The public interface was set to use the Avaya SBCE default media port range of to Navigate to Device Specific Setting Media Interface from the left-side menu to configure Media Interfaces; one for internal and one for external. Under Devices, select the device being managed, which was named ASBCE-1 in the sample configuration (not shown). Select Add to add a media interface. 55 of 69

56 Enter an appropriate Name for the Media Interface facing the enterprise and select the inside, private IP Address for the Avaya SBCE from the IP Address drop-down menu. In the sample configuration, Media_Inside was chosen as the name, and the inside IP Address of the SBC is For the Port Range, the IP Office default media port range of to is shown. Click Finish. An external Media Interface facing the network was similarly created with the name Media_Outside. The outside IP Address of the SBC ( ) was selected from the dropdown menu and the Port Range setting was left at the Avaya SBCE default value of to The resultant Media Interface configuration used in the sample configuration is shown below. 56 of 69

57 6.12. Device Specific Settings Signaling Interface Navigate to Device Specific Setting Signaling Interface from the left-side menu to configure Signaling Interfaces; one for internal and one for external. Under Devices, select the device being managed, which was named ASBCE-1 in the sample configuration (not shown). Select Add to add a signaling interface. In the Add Signaling Interface screen, enter an appropriate Name (e.g., Sig_Inside) for the inside interface, and choose the private, inside IP Address of the Avaya SBCE from the IP Address drop-down menu. Enter 5060 for TCP Port since TCP port 5060 is used between IP Office and the Avaya SBCE in the sample configuration. Click Finish. 57 of 69

58 An external Signaling Interface facing the network was similarly created with the name Sig_Outside. Select the outside, public IP Address of the Avaya SBCE ( ) from the drop-down menu. Note that 5060 was specified as the UDP Port since UDP was used between the Avaya SBCE and the Sotel Systems network. The following screen shows the Signaling Interfaces defined for the sample configuration Device Specific Settings End Point Server Flows End Point Server Flows combine the previously defined profiles into an outgoing flow from the Call Server (IP Office) to the Trunk Server (service provider network) and an incoming flow from the Trunk Server to the Call Server. This configuration ties all the previously entered information together so that calls can be routed from IP Office to the service provider network and vice versa. Select Device Specific Setting End Point Flows from the left-side menu to configure End Point Flows. 58 of 69

59 Under Devices, select the device being managed, which was named ASBCE-1 in the sample configuration (not shown). Select the Server Flows tab and select Add. The following screen shows the flow named Avaya being added to the sample configuration. This flow uses the interfaces, policies, and profiles defined in previous sections. Note the Routing Profile selection which is the reverse route of the flow. Click Finish. Once again, select the Server Flows tab and select Add. 59 of 69

60 The following screen shows the flow named ServiceProvider being added to the sample configuration. This flow uses the interfaces, policies, and profiles defined in previous sections. Note the Routing Profile selection which is the reverse route of the flow. Click Finish. The following screen summarizes the Server Flows configured in the sample configuration. 60 of 69

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62 7. Sotel Systems SIP Trunking Configuration Sotel Systems is responsible for the configuration of Sotel Systems SIP Trunking. The customer will need to provide the IP address used to reach the Avaya IP Office at the enterprise. Sotel Systems will provide the customer the necessary information to configure the Avaya IP Office SIP connection to Sotel Systems including: IP address / FQDN of the Sotel Systems SIP proxies. Supported codecs DID numbers All IP addresses and port numbers used for signaling or media that will need access to the enterprise network through any security devices 8. Verification Steps This section provides verification steps that may be performed in the field to verify that the solution is configured properly. These steps are for informational purposes and were not taken from the specific compliance test performed for these Application Notes. As such, IP addresses and phone numbers differ from those used in the actual compliance test. The verification steps listed here however are still relevant and valid for the configuration used in these Application Notes System Status The System Status application is used to monitor and troubleshoot IP Office. Use the System Status application to verify the state of the SIP trunk. System Status can be accessed from Start Programs IP Office System Status. Or by opening an Internet browser and type the URL: where ipaddress is the IP address of the Avaya IP Office LAN1 interface. Click on System Status to launch the application. 62 of 69

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64 The following screen shows an example Logon screen. Enter the IP Office IP address in the Control Unit IP Address field, and enter an appropriate User Name and Password. Click Logon. Select the SIP line under Trunks from the left pane. On the Status tab in the right pane, verify the Current State is Idle for each channel. 64 of 69

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66 Select the Alarms tab and verify that no alarms are active on the SIP line Monitor The Monitor application can also be used to monitor and troubleshoot IP Office. Monitor can be accessed from Start Programs IP Office Monitor. The application allows the monitored information to be customized. To customize, select the button that is third from the right in the screen below, or select Filters Trace Options. The following screen shows the SIP tab, allowing configuration of SIP monitoring. In this example, the SIP Rx and SIP Tx boxes are checked. All SIP messages will appear in the trace with the color blue. To customize the color, right-click on SIP Rx or SIP Tx and select the desired color. 66 of 69

67 As an example, the following shows a portion of the monitoring window for an outbound call from extension 209, whose DID is , calling out to the PSTN via the Sotel Systems IP Trunking Service. The telephone user dialed Conclusion These Application Notes describe the configuration necessary to connect Avaya IP Office 8.1 to Sotel Systems SIP Trunking service. Sotel Systems SIP Trunking is a SIP-based Voice over IP solution for customers ranging from small businesses to large enterprises. It provides a flexible, cost-saving alternative to traditional hardwired telephony trunks. Sotel Systems SIP Trunking passed compliance testing. Please refer to Section 2.2 for any exceptions. 67 of 69