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Document Overview

This configuration guide provides instructions for Sonus SBC Edge (1000/2000) Series (Session Border Controller) when deployed in support of Microsoft® Skype for Business® 2015 Server (SFB2015)A secondary goal is to demonstrate how the SBC attaches 3rd party SIP-based non-SFB2015 clients into the SFB2015 environment, including the offer to these clients of network redundancy to overcome typical failure scenarios. In this paper, Polycom® VVX® SIP endpoints are configured to assume this non-SFB2015 endpoint role.

This configuration guide supports features identified on Microsoft Technet.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flows between SBC 1000, its subtended clients (SIP-based endpoints, TDM/FXx endpoints/trunks, etc.) and the SFB2015 infrastructureWhile all the examples refer to the SBC 1000, please note the instructions and resulting behavior are also applicable to the SBC 2000. 

Audience

This technical document is intended for telecommunications engineers with the purpose of configuring both the Sonus SBC Edge, example SBC subtended endpoints, and the Skype for Business infrastructure. There will be steps that require navigating third-party as well as the Sonus SBC 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
*Tenor AF P108-09-21
Third-party
MicrosoftMicrosoft Skype for Business 2015 (Skype 2015) Mediation 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
VentaFaxVentaFax7.6.243.597 I

*Tenor GW is there for demonstration purposes. The fax machine may be directly connected to the SBC 1000's FXS port if desired.


Reference Configuration

The following reference configuration shows connectivity between Skype for Business infrastructure and Sonus SBC 1000.

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 SFB2015 client: PSTN -> SBC -> SFB2015 -> SFB2015 client
  • Call from PSTN to analog phones*: PSTN -> SBC -> SFB2015 -> SBC -> analog phones*
  • Calls from PSTN to fax: PSTN -> SBC -> SFB2015 -> SBC -> Tenor GW -> Fax
  • Calls from fax to PSTN: Fax -> Tenor GW -> SBC -> SFB2015 -> SBC -> PSTN
  • Calls from analog phones* to PSTN: *analog phone -> SBC -> SFB2015 -> SBC -> PSTN
  • Calls from analog phones* to SFB2015 client:  analog phones* -> SBC -> SFB2015 -> SFB2015 client
  • Calls from analog phones* to analog phones*: analog phones* -> SBC -> SFB2015 -> SBC-> analog phones*
  • Calls from SFB2015 client to PSTN: SFB2015 client -> SFB2015 -> SBC -> PSTN
  • Calls from SFB2015 client to analog phones*:  SFB2015 client -> SFB2015 -> SBC -> analog phones*
  • Failover scenario (SFB2015 unavailable): Call from PSTN to analog phones*: PSTN -> SBC -> analog phones*
  • Failover scenario (SFB2015 unavailable): Calls from PSTN to fax: PSTN -> SBC -> Tenor GW -> Fax
  • Failover scenario (SFB2015 unavailable): Calls from fax to PSTN: Fax -> Tenor GW -> SBC  -> PSTN
  • Failover scenario (SFB2015 unavailable): Calls from analog phones* to PSTN: analog phones* -> SBC -> PSTN
  • Failover scenario (SFB2015 unavailable): Calls from analog phones* to analog phones*: analog phones* -> SBC -> analog phones*

* 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 for Business 2015 Server Configuration

 

The following configuration steps are provided to configure SFB2015 to interoperate with the Sonus SBC 1000General SFB2015 environment variables 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 .
  1. PSTN Gateway
  2. Voice Policy
  3. PSTN Usage
  4. Route
  5. Trunk Configuration
  6. Analog Devices Feature Configuration

1. PSTN Gateway

Configure the PSTN Gateway using the following configuration screens:

Figure : Define a new IP/PSTN Gateway

Figure : Define FQDN

 

*192.168.10.10 is the IP address of the Logical Interface assigned to the SFB2015 Signaling Group of the SBC 1000.

Figure : Define IP Address Type

Figure : Define Root Trunk

 

2. Voice Policy

Select Control Panel > Voice Routing > Voice Policy to access the Voice Policy configuration screen. 

Figure : Voice Policy

 

3. PSTN Usage

Select Control Panel > Voice Routing > PSTN Usage to access the PSTN Usage configuration screen.

Figure : PSTN Usage

 

4. Route

Select Control Panel > Voice Routing > Route to access the Route configuration screen.

Figure : Route

 

5. Trunk Configuration

Select Control Panel > Voice Routing > Trunk Configuration to access the trunk configuration screen.

Figure : Trunk Configuration

 

6. Analog Devices Feature Configuration

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

Figure : Windows Power Shell

 

To create new instance of the Analog Device that you can manage with the Skype server, use the New-CsAnalogDevice command. The following are examples to create the Analog Phone and Fax:

 Analog Device commands

PS C:\Users\administrator.SKYPE2015> New-CsAnalogDevice -LineUri tel:+1222111333 -DisplayName "Poly1" -RegistrarPool fe.
skype2015.sonusnet.com -AnalogFax $False -Gateway 10.35.177.230 -OU "OU=Contacts,DC=SKYPE2015,DC=SONUSNET,DC=COM"


Identity : CN=Poly1,OU=Contacts,DC=SKYPE2015,DC=SONUSNET,DC=COM
VoicePolicy :
VoiceRoutingPolicy :
RegistrarPool : fe.skype2015.sonusnet.com
Gateway : 10.35.177.230
AnalogFax : False
Enabled : True
SipAddress : sip:5522040b-3c91-4331-b351-ccadbe965035@skype2015.sonusnet.com
LineURI : tel:+1222111333
DisplayName : Poly1
DisplayNumber :
ExUmEnabled : False

 

PS C:\Users\administrator.SKYPE2015> New-CsAnalogDevice -LineUri tel:+1222111000 -DisplayName "Fax1" -RegistrarPool fe.s
kype2015.sonusnet.com -AnalogFax $True -Gateway 10.35.177.230 -OU "OU=Contacts,DC=SKYPE2015,DC=SONUSNET,DC=COM"


Identity : CN=Fax1,OU=Contacts,DC=SKYPE2015,DC=SONUSNET,DC=COM
VoicePolicy :
VoiceRoutingPolicy :
RegistrarPool : fe.skype2015.sonusnet.com
Gateway : 10.35.177.230
AnalogFax : True
Enabled : True
SipAddress : sip:a5d4f0e3-6c02-4787-867e-e5101b32202f@skype2015.sonusnet.com
LineURI : tel:+1222111000
DisplayName : Fax1
DisplayNumber :
ExUmEnabled : False

 

The preceding commands will create an Analog Device with Analog Phone and Fax functions. The following list describes the parameters:

  • LineUriPhone number for the analog device. The line Uniform Resource Identifier (URI) should be specified by using the E.164 format, and be prefixed by the "TEL:" prefix. 
  • DisplayName -  Configures the Active Directory display name of the analog device.
  • RegistrarPoolFully qualified domain name (FQDN) of the Registrar pool where the contact object should be homed.
  • AnalogFaxSet to True ($True) if the analog device is a fax machine. Set to False ($False) if the device is not a fax machine.
  • GatewayIP address of the PSTN gateway to be used by the analog device.
  • OUDistinguished name of the Active Directory organizational unit (OU) where the contact object should be located. 

Sonus SBC 1000/2000 Configuration

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

1. SIP Profile

Select Settings > SIP > SIP Profiles

SIP Profiles control how the Sonus SBC Edge 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 Edge for this testing effort:

 

Figure : SIP Profiles

 

2. SIP Server Tables 

Select Settings > Security > SIP Server Tables

SIP Server Tables contain information about the SIP devices connected to the Sonus SBC Edge. 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 : Skype

 

Figure : PSTN


3. 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 Edge in this testing effort and are shown for reference only:

 

Figure : Voice Codec G711 A-Law

 

Figure : Voice Codec G711 U-Law

 

4. 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

 

5. 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 to Analog


 

Figure : From PSTN to SfB


 

Figure : From Analog to PSTN

Figure : From Analog to SfB

Figure : From Skype to Analog

Figure : From Skype to PSTN


6. Cause Core Reroutes 

Select Settings > Telephony Mapping Tables

Terminating ISDN calls return a Q.850 Cause Code when they end. These codes can be used to determine whether or not to reroute the call to another signalling group. A Cause Code Reroute table contains one or more Q.850 Cause Codes that when matched, triggers a reroute.

 

Figure : Cause Code Reroutes Table


7. 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 Skype to Analog and PSTN

Figure : From Analog

8. 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 addressesThis registrar feature is used by the subtended Polycom VVX SIP-based endpoints. 


Figure : Registrar

9. 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

10. 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. This is 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 Skype

Figure : To/From Analog

Figure : To/From PSTN

 

VVX Phone Configuration

The following snapshots show the Polycom VVX endpoint configuration used to accomplish SIP and RTP-based communications to the SBC.  Recall the Polycom VVX phones, while they interact with the SBC through SIP and RTP, are for the purposes of the SFB2015 infrastructure, the equivalent of analog (for example, FXS-based) endpoints.

Figure : SIP Server

Figure : Authentication and Identification

 

1. Call to/from PSTN to SFB2015 client: PSTN <-> SBC <-> SFB2015 <-> SFB2015 client

 

Figure : Call to/from PSTN to Skype client: PSTN <-> SBC <-> Skype <-> Skype client


2.Call to/from PSTN to "analog" phones: PSTN <-> SBC <-> SFB2015 <-> SBC <-> "analog" phones

 

Figure : Call to/from PSTN to "analog" phones: PSTN <-> SBC <-> SFB2015 <-> SBC <-> "analog" phones

 


3.Calls to/from PSTN to fax: PSTN <-> SBC <-> SFB2015 <-> SBC <-> Tenor GW <-> Fax

 

Figure : Calls to/from PSTN to fax: PSTN <-> SBC <-> SFB2015 <-> SBC <-> Tenor GW <-> Fax

 


4.Calls to/from "analog" phone to SFB2015 client:  "analog" phone <-> SBC <-> SFB2015 <-> SFB2015 client

 

Figure : Calls to/from "analog" phone to SFB2015 client:  "analog" phone <-> SBC <-> SFB2015 <-> SFB2015 client


5.Failover scenario (SFB2015 unavailable): Call to/from PSTN to "analog" phones: PSTN -> SBC -> "analog" phones

 

Figure : Failover scenario (SFB2015 unavailable): Call to/from PSTN to "analog" phones: PSTN -> SBC -> "analog" phones

 


6.Failover scenario (SFB2015 unavailable): Calls to/from PSTN to fax: PSTN -> SBC -> Tenor GW -> Fax

 

Figure : Failover scenario (SFB2015 unavailable): Calls to/from PSTN to fax: PSTN -> SBC -> Tenor GW -> Fax

 

 

Test Results

 

S.NoProcedureObservationResultComment
1Call From PSTN to SFB2015 client: PSTN -> SBC -> SFB2015 -> SFB2015 client Pass 
2Call from PSTN to "analog" phones: PSTN -> SBC -> SFB2015 -> SBC -> "analog" phones Pass 
3Calls from PSTN to fax: PSTN -> SBC -> SFB2015 -> SBC -> Tenor GW -> Fax Pass 
4Calls from fax to PSTN: Fax ->Tenor GW -> SBC -> SFB2015 -> SBC -> PSTN Pass 
5Calls from "analog" phone to PSTN: "analog" phone -> SBC -> SFB2015 -> SBC -> PSTN Pass 
6Calls from "analog" phone to SFB2015 client:  "analog" phone -> SBC -> SFB2015 -> SFB2015 client Pass 
7Calls from "analog" phone to "analog" phone: "analog" phone -> SBC -> SFB2015 -> SBC -> "analog" phone Pass 
8Calls from SFB2015 Client to PSTN: SFB2015 client -> SFB2015 -> SBC - >PSTN Pass 
9Calls from SFB2015 to "analog" phones:  SFB2015 client -> SFB2015 -> SBC -> "analog" phone Pass 
10Failover scenario (SFB2015 unavailable): Call from PSTN to "analog" phones: PSTN -> SBC -> "analog" phones PassSBC 1000 assumes the role of the backup SIP server, and routes the call to the analog phone directly based on its own routing tables
11Failover scenario (SFB2015 unavailable): Calls from PSTN to fax: PSTN -> SBC -> Tenor GW -> Fax PassSBC 1000 assumes the role of the backup SIP server, and routes the call to the Fax directly based on its own routing tables
12Failover scenario (SFB2015 unavailable): Calls from fax to PSTN: Fax -> Tenor GW -> SBC  -> PSTN PassSBC 1000 assumes the role of the backup SIP server, and routes the call to the PSTN directly based on its own routing tables
13Failover scenario (SFB2015 unavailable): Calls from "analog" phone to PSTN: "analog" phone -> SBC -> PSTN PassSBC 1000 assumes the role of the backup SIP server, and routes the call to the PSTN directly based on its own routing tables
14Failover scenario (SFB2015 unavailable): Calls from "analog" phone to "analog" phone: "analog" phone -> SBC -> "analog" phone PassSBC 1000 assumes the role of the backup SIP server, and routes the call to the analog phone directly based on its own routing tables

Conclusion

This Application Note describe the configuration steps required for Sonus SBC Edge to successfully interoperate with with SFB2015. The document also successfully demonstrates how the SBC provides call services to subtended clients in the the event of a network failure or service disruption regarding the SFB2015 environment. All feature and serviceability test cases were completed and passed with the exceptions and observations noted in Test Results.