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Document Overview

This configuration guide provides instructions for activating and validating the Presence Reporting feature for non-Skype for Business® 2015 (SFB2015) endpoints on the Sonus SBC Edge (1000/2000) (Session Border Controller). Please refer to other available configuration guides for general guidelines regarding SfB 2015 and SBC configuration setup and call flows.

This configuration guide supports features identified in Microsoft Technet.

Introduction

The interoperability compliance testing focuses on verifying the presence reporting associated with SBC 1000 non-SfB 2015 subtended endpoints while engaged/not engaged in calls in a SfB 2015 environmentWhile all the examples refer to the SBC 1000, please note the instructions and resulting behavior are also applicable to the SBC 2000. 

Document History

 

DateNameComment
28.12.2016Ilija DameskiInitial Draft
   

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC Edge and the Skype for Business infrastructure. There will be steps that require navigating third-party as well as the Sonus SBC Edge Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary. 

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this guide.

 

Requirements

The following equipment and software were used for the sample configuration provided:

Table : Test Equipment and Software

Vendor

Equipment

Software Version

Sonus NetworksSBC 1000 V6.0.0build435
Third-party
MicrosoftMicrosoft Skype for Business 2015 (Skype 2015) Server6.0.9319.0
PolycomPolycom VVX310 SIP Phone5.4.0.10182
PolycomPolycom VVX410 SIP Phone5.4.0.10182
PolycomPolycom VVX500 SIP Phone5.4.0.10182
PolycomPolycom VVX600 SIP Phone5.4.0.10182

Reference Configuration

The following reference configuration shows connectivity between Skype for Business infrastructure, the Sonus SBC 1000, and subtended non-SfB 2015 endpoints (Polycom VVX SIP phones) for which the SBC 1000 will publish presence related information.

Figure : Reference Configuration Topology

Support

For any questions regarding this document or the content herein, please contact your maintenance and support provider.

 

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://support.sonus.net/display/UXDOC61/Viewing+Licenses for a description of licensable features, and for follow-on references regarding license acquisition and submission.

 

Skype Server 2015 Configuration for Presence Feature

 

The following configuration steps are provided to configure Skype for Business Server 2015 to interoperate with the Sonus SBC 1000/2000. Skype for Business Server 2015 environment should have been setup prior to undertaking these specific steps according to the direction posted at https://technet.microsoft.com/library/gg398616(v=ocs.16).aspx .

 

Although not shown, the phone number +1222111333 is defined in AD (Active Directory(r)) as attribute homePhone for a Skype user (msRTCSIP-PrimaryUserAddress = prgskype3@example.com). It is this relationship that allows the SBC 1000 to publish presence for the "analog" phone (i.e. the Polycom VVX endpoint) assigned the +1222111333 number.


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:

 Analog Device commands

Get-CsSite

#Create the application pool

New-CsTrustedApplicationPool -Identity powder230.sfb.tst -Registrar cspool.sfb.tst -Site 1

#Create the trusted application

New-CsTrustedApplication -ApplicationId powder230.sfb.tst -TrustedApplicationPoolFqdn powder230.sfb.tst -Port 5061

Enable-CsTopology

 

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".

Figure : View of the user status when in call

Sonus SBC 1000/2000 Configuration

The following steps provide an example of how to configure Sonus SBC 1000/2000:

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:

 

Figure : 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

Figure : Sonus SBC Certificate

 

3. TLS Profiles

Select Settings > Security > TLS Profiles

After the Sonus SBC 1000/2000 obtains the required certificates, configuration of several options/attributes on both the server and client is necessary before TLS can employ the certificate(s) in establishing a secure connection. The attributes are configured in TLS profiles. Attributes include, but are not limited to, such things as Client Ciphers, and inactivity timeouts.


 

Figure : 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.

Figure : PSTN

Figure : 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.


Figure : Node-Level SIP Settings

6. Authentication and Directory Services

Select Settings > Auth and Directory Services > Configuration


Figure : Auth and Directory Services - Configuration

 

Select Settings > Auth and Directory Services > Domain Controllers


Figure : 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:

 

Figure : Voice Codec G711 A-Law

 

Figure : 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.

 

Figure : Media Lists

 

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.

 

Figure : From PSTN

Figure : From Polycom

Figure : Presence



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.


Figure : Action Configuration

 

11Actions 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 entryWe use Actions Sets to enable the presence reporting feature.


Figure : 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).

 

Figure : From PSTN

Figure : 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 addressesThe registration feature is required to properly support the Polycom VVX endpoints from the SBC 1000.


Figure : Registrar

14. Local/Pass-Thru Auth Table

Select Settings > SIPLocal/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.


Figure : 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.

 

Figure : To/From Polycom

Figure : 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. 


Figure : SIP Server

Figure : Authentication and Identification



Test Results

 

S.NoProcedureObservationResultComment
1Call from PSTN to "analog" phones: PSTN -> SBC -> "analog" phonesPresence information for the analog phones when the call was established was updated from "free" to "busy". When the call was ended, the presence for was set to "free".Pass 
2Calls from "analog" phone to PSTN: "analog" phone -> SBC -> PSTNPresence information for the analog phones when the call was established was updated from "free" to "busy". When the call was ended, the presence for was set to "free".Pass 
3Calls from "analog" phone to "analog" phone: "analog" phone -> SBC -> "analog" phonePresence information for the analog phones when the call was established was updated from "free" to "busy". When the call was ended, the presence for was set to "free".Pass 

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.

Figure : 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.

Figure : 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

Figure : Analog phone status after the call

Conclusion

These Application Notes describe the configuration steps required for Sonus SBC 1000/2000 to successfully interoperate with Skype for Business 2015. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.