Sonus SBC 1000/2000 V6.0.0 Registration IOT Skype for Business 2015 Presence Feature Application Note

Third-Party Product Features

The following call flows are supported:

• Call from PSTN to analog phones*: PSTN → SBC → analog phones*
• Calls from analog phones* to PSTN: analog phones* → SBC → PSTN
• Calls from analog phones* to analog phones*: analog phones* → SBC → analog phones*
• Presence Feature: Presence status updates for analog phones* on SFB2015 clients
  
  
  

*Please note the analog phones are the Polycom VVX SIP-based phones listed in Table 1, as opposed to an FXS based phone. These Polycom endpoints are considered as "analog" clients from the perspective of the Skype for Business Server 2015, as documented at https://technet.microsoft.com/en-us/library/gg398314(v=ocs.14).aspx

Verify License

The following SBC 1000 licensable features are required for the documented scenarios to work as described:

• SIP Calls (minimum of 1 license)
• SIP Registrations (minimum of 1 license)

Please refer to https://doc.rbbn.com/display/UXDOC61/Viewing+Licenses for a description of licensable features, and for follow-on references regarding license acquisition and submission.

In Skype Server 2015, start the Windows Power Shell (point to the Windows Start menu, click All Programs, and then click Windows Power Shell).

To create new instance of presence application and trusted application pool you can use the following command:

The preceding command will create presence application and trusted application pool. Parameters for this command are:

• Identity - FQDN of the SBC providing presence updates to Skype Server.
• Registrar - FQDN of the Registrar service for the pool.
• Site - Site Id obtained in Get-CsSite
• ApplicationId - The name of the application.
• TrustedApplicationPoolFqdn - The FQDN of the trusted application pool created in Create a Trusted Application Pool.
• Port - The port number on which the application will run (5061).

Once the commands are accepted, the SFB2015 structure will recognize and support the reporting of presence associated with the phone number +1222111333, currently set to display as user "PrgSkype3".

View of the user status when in call

1. SIP Profile

Select Settings > SIP > SIP Profiles

SIP Profiles control how the Sonus SBC 1000/2000 communicates with SIP devices. These control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags. The following figure shows the default SIP profile used for the SBC 1000/2000 for this testing effort:

SIP Profiles

2. SBC Certificates 

Select Settings > Security > SBC Certificates > Sonus Certificate

By default, after having initialized the Sonus SBC 1000/2000 system for the first time, or after a factory reset, the Sonus SBC 1000/2000 system is pre-configured with a Self-signed Server Certificate.

The process of installing a new Signed Certificate on the Sonus SBC 1000/2000 comprises two general steps which must be followed in this specific order:

1. Generate a Certificate Signing Request (CSR)
2. Obtain and Import the Signed Server Certificate
   
   

Sonus SBC Certificate

 

3. TLS Profiles

Select Settings > Security > TLS Profiles

Skype

 

4. SIP Server Tables 

Select Settings > Security > SIP Server Tables

SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000. The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting.

PSTN

Skype Presence Server

5. Node-Level SIP Settings

Select Settings > SIP > Node-Level SIP Settings

The Node-Level SIP Settings feature enables SIP options to be configured that apply to the SBC 1000/2000 at the node level.

Node-Level SIP Settings

6. Authentication and Directory Services

Select Settings > Auth and Directory Services > Configuration

Auth and Directory Services - Configuration

Select Settings > Auth and Directory Services > Domain Controllers

7. Media Profile

Select Settings > Media > Media Profiles

Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidth requirements at the expense of voice quality. The following figures are the media profiles of the voice codecs used for the SBC 1000/2000 in this testing effort and are shown for reference only:

Voice Codec G711 A-Law

 

Voice Codec G711 U-Law

 

8. Media List

Select Settings > Media > Media List

The Media List shows the selected voice and fax compression codecs and their associated settings.

9. Transformation Table

Select Settings > Transformation

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For example, Transformation Tables can convert a public PSTN number into a private extension number, or into a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table, and are sequentially selected from there. In addition, Transformation tables are configurable as a reusable pool that Action Sets can reference.

From PSTN

From Polycom

10. Action Configuration 

Select Settings > Call Actions > Action Configuration

Action Configurations are the individual steps (actions) that make up an Action Set (Action Sets allow you to perform additional actions before routing a call). Through Action Configuration, you create an Action (such as Route Call, Send Connect, Detect CNG, etc.) and assign a Routing table to perform that action.

Action Configuration

11. Actions Sets 

Select Settings > Call Actions > Actions Sets

Action Sets allow you to perform additional actions before routing a call; they provide the capability to perform a high degree of number normalization before engaging the call router. They are comprised of a number of Actions, which must first be defined in Action Configurations. Once created, an Action Set Table is a sequence of steps to follow to route a call. To use an Action Set, select it from the Action Set Table field in a Signaling Group entry. We use Actions Sets to enable the presence reporting feature.

Action Sets

12. Call Routing Table

Select Settings > Call Routing Table

Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls will be carried, and also how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).

From PSTN

From Polycom

13. Registrar

Select Settings > SIP > Local Registrars

SIP provides a registration function that allows users to upload their current locations for use by proxy servers. Registration creates bindings in a location service for a particular domain that associates an address-of-record URI with one or more contact addresses. The registration feature is required to properly support the Polycom VVX endpoints from the SBC 1000.

Registrar

14. Local/Pass-Thru Auth Table

Select Settings > SIP > Local/Pass-through Authorization Tables

Local Pass-through Tables contain entries with information about SIP endpoints. The SBC Edge uses this information to challenge SIP request messages such as REGISTER. It is used in the SIP Signaling Group when the Challenge Request is enabled.

Local/Pass-Thru Auth Table

15. Signaling Groups

Select Settings > Signaling Groups

Signaling Groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. These groups are the entity to which calls are routed, as well as the location from which Call Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, this will specify protocol settings and link to server, media, and mapping tables.

To/From Polycom

To/From PSTN

VVX Phone Configuration

Figures 25 and 26 present the configuration of the Polycom VVX phone assigned the +1222111333 phone number, associated to the prgskype3@example.com Skype user in AD. For the VVX phones, we must undertake the configuration of the SIP server and Outbound Proxy towards the SBC 1000 as the "home" server. Additionally, we need to configure the authentication and identification information as configured in the Local/Pass-Thru Auth Table of the SBC 1000. 

SIP Server

Authentication and Identification

Test Results

Status of the analog phone during test cases execution 

Status of the analog phone on SFB2015 clients when not engaged in a call , i.e. before test case 1 is run.

Analog phone status before the call

Status of the analog phone on SFB2015 clients when engaged in a call , i.e. while test case 1 was running.

Analog phone status during the call

Status of the analog phone on SFB2015 clients when the call was over, i.e. after test case 1 was completed

Analog phone status after the call