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Document Overview

This Application Note is a configuration guide for the Sonus Session Border Controller (SBC) 1000/2000 Series when connecting to Skype for Business 2015 (Skype 2015) and Pure IP SIP Trunk.

This configuration guide supports features described on the Microsoft Technet https://technet.microsoft.com/ website.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Sonus SBC 1000/2000, Skype 2015 and Pure IP SIP Trunk.

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. Navigating the third-party as well as the Sonus SBC Command Line Interface (CLI) will be required. Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and any needed troubleshooting.

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 : Requirements

 

Equipment

Software Version

Sonus Networks

SBC 2000

V6.1.1build459

Tenor AFM200P108-09-26

Third-party Equipment

 

Microsoft Skype for Business 2015 (Skype 2015) Mediation Server 6.0.9319.0
Polycom CX600 SIP Phone

6.3037

VentaFax

4.0.7577.44455


Reference Configuration

The following reference configuration shows connectivity between Pure IP SIP Trunk, Skype 2015 and Sonus SBC 1000/2000.

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 testing was executed with the Pure IP test plan. The following features were tested:

  • SIP Options
  • Initial Calls To/From PSTN
  • Codec Negotiations
  • Incomplete Call Attempts
  • DTMF Tone Support
  • PSTN Numbering Plans
  • Calling Name / Number Blocking
  • Supplementary Features – Call Hold, Forward, Transfer, Conference 
  • Fax Support
  • Stability and Maintenance

Verify License

SIP Calls

 

Skype for Business 2015 Configuration

The following new configurations are included in this section:

  1. PSTN Gateway
  2. Voice Policy
  3. PSTN Usage
  4. Route
  5. Trunk Configuration

1. PSTN Gateway

Select Topology Builder > Shared Components > PSTN Gateways

Figure : Define a new IP/PSTN Gateway

 

Figure : Define FQDN

 

Figure : Define IP Address

 

Figure : Define Root Trunk

 

2. Voice Policy

Select Control Panel > Voice Routing > Voice Policy

Figure : Edit Voice Policy


 

3. PSTN Usage

Select Control Panel > Voice Routing > PSTN Usage

Figure : View PSTN Usage

 

4. Route

Select Control Panel > Voice Routing > Route

Figure : Edit Voice Route

 

 

5. Trunk Configuration

Select Control Panel > Voice Routing > Trunk Configuration

Figure : Edit Trunk Configuration

 

Sonus SBC 1000/2000 Configuration

The following steps provide an example of how to configure the 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. SIP Server

Select Settings > SIP > 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 : Skype

 

Figure : Fax

 

Figure : Pure IP

 
 

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 1000/2000 in this testing effort and are shown for reference only:

Figure : Voice Codec G711 A-Law

 

Figure : Voice Codec G711 U-Law

 

Figure : Voice Codec G729

 

Figure : T.38

 
 

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 List

 

5. Transformation Table

Select Settings > Transformation

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For example, transformations 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 is sequentially selected from there. In addition, Transformation tables are configurable as a reusable pool that Action Sets can reference.

Figure : From Pure IP

 

Figure : From Pure IP Fax

 

Figure : From Skype to Pure IP

 

6. 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 Pure IP

Figure : From Skype to Pure IP

 


7. 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 : Internal Side

 

Figure : External Side

 

 

Test Results

 

Table : Test Results

S.NoProcedureObservationResultComment
SIP Options
1.1.1Validate OPTIONS messages sent  Pass 
1.1.2Validate OPTIONS messages received Pass 
Initial Calls To/From PSTN
1.2.1Inbound call from a PSTN phone to an extension.  Hang-up at called party. Wait for calling party to disconnect. Pass 
1.2.2Inbound call from a PSTN phone to an extension.  Hang-up at calling party (PSTN phone). Wait for called party to disconnect. Pass 
1.2.3Outbound call from an extension to a PSTN phone.  Hang-up at called party (PSTN phone). Wait for calling party to disconnect. Pass 
1.2.4Outbound call from an extension to a PSTN phone.  Hang-up at calling party (extension). Wait for called party to disconnect. Pass 
Codec Negotiations
1.3.1

Make 3 outbound test calls:

-       one offering G711A only,

-       one offering G711U only,

-       one offering G729 only. 

 Pass 
1.3.2

Make 3 inbound test calls:

-       one offering G711A only,

-       one offering G711U only,

-       one offering G729 only. 

 Pass 
Incomplete Call Attempts
1.4.1Inbound call from a PSTN phone to an extension.  Hang-up before far-end answers. Pass 
1.4.2Outbound call from an extension to a PSTN phone.  Hang-up before far-end answers. Pass 
1.4.3No Answer of inbound call from a PSTN phone to an extension.   Pass 
1.4.4No Answer of outbound call from an extension to a PSTN phone Pass 
1.4.5Inbound call from a PSTN phone to an extension that is “Busy”.   N/ASkype for Business doesn’t send 486 Busy
1.4.6

Outbound call from an extension to a PSTN phone that is “Busy”.  

 Pass 
1.4.7Outbound call from an extension to an invalid PSTN number. Pass 
DTMF Tone Support
1.5.1Outbound call to PSTN number that requires DTMF input Pass 
1.5.2Inbound call to number that requires DTMF input Pass 
PSTN Numbering Plans
1.6.1Outbound Toll-Free Call Pass 
1.6.2Outbound Local Call Pass 
1.6.3Outbound International Calls Pass 
1.6.4Local Directory Assistance Call or special number Pass 
1.6.5Emergency Calls  N/AUpon agreement this test is skipped not to cause any issues with local PSP. 
Calling Name / Number Blocking
1.7.1Inbound Calling Party Number (CPN) Block from PSTN phone Pass 
1.7.2Outbound Calling Party Number (CPN) Block from phone Pass 
Supplementary Features – Call Hold, Forward, Transfer, Conference
1.8.1Inbound Call – PBX Hold and Resume – Short Duration Pass 
1.8.2Outbound Call – PBX Hold and Resume – Short Duration Pass 
1.8.3Inbound Call – PSTN Hold and Resume – Short Duration Pass 
1.8.4Outbound Call – PSTN Hold and Resume – Short Duration Pass 
1.8.5Call - PBX Hold and Resume – Long Duration that exceeds the SIP session timers (~10 min) Pass 
1.8.6Call Forward – All of inbound call from PSTN to another extension Pass 
1.8.7Call Forwarding – Busy / Don’t Answer of inbound call from PSTN to another extension Pass 
1.8.8Call Forwarding Off Net over SIP Trunk - Inbound call from PSTN to extension is forwarded to another PSTN endpoint. Pass 
1.8.9Blind Call Transfer (xfer before answer) of inbound PSTN call transferring to internal extension Pass 
1.8.10Blind Call Transfer (xfer before answer) of inbound PSTN call transferring to another PSTN endpoint. Pass 
1.8.11Consultative Call Transfer (xfer after answer) of inbound PSTN call transferring to internal extension Pass 
1.8.12Consultative Call Transfer (xfer after answer) of inbound PSTN call transferring to another PSTN endpoint Pass 
1.8.13Consultative Call Transfer (xfer after answer) of outbound PSTN call transferring to internal extension Pass 
1.8.14Conference of inbound call  Pass 
1.8.15Conference outbound call (with internal users) Pass 
Fax Support
1.9.1

Incoming fax using T.38 

 Pass 
1.9.2

Outbound fax using T.38 

 Pass 
1.9.3Incoming fax using G711  Pass 
1.9,4Outbound fax using G711  Pass 
Stability and Maintenance
1.10.1Long Duration Calls Pass 
1.10.2Incoming & Outgoing Call – SIP Trunk Signaling Failure Pass 

 

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.