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Document Overview

This configuration guide is for Sonus SBC 1000/2000 series (Session Border Controller) when connecting to Skype for Business 2015 and TELUS SIP Trunking using IP Authentication.

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

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flows between Sonus SBC 1000/2000 and Skype for Business 2015.

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. 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 : Requirements

 

Equipment

Software Version

Sonus Networks 

SBC 2000

V6.0.1build441

Tenor AFM200P108-09-26
Applicable SBC versions V6.0.XbuildXXX

Third-party Equipment

 

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

4.0.7577.44455

VentaFax

7.6.243.597 I


Reference Configuration

The following reference configuration shows connectivity between third-party and Sonus SBC 1000/2000.

Figure : Connectivity Between Third-Party and Sonus SBC 2000

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 Telus test plan. The following features were tested for PSTN, BVOIP, Mobile, VoLTE and HGHP clients:

  • Basic originated and terminated calls 
  • Basic inbound/outbound call 
  • Basic inbound/outbound call with privacy 
  • Hold and resume 
  • Call Transfer (Blind transfer)
  • Call Transfer (Consult transfer)
  • Call Forwarding Unconditional 
  • Call Forwarding Busy
  • Call Forwarding Don’t Answer
  • Voicemail 
  • Conference call 
  • Long calls
  • FAX 
  • DTMF
  • International calls

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

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

Control Panel > Voice Routing > Voice Policy

Figure : Edit Voice Policy


 

3. PSTN Usage

Control Panel > Voice Routing > PSTN Usage

Figure : View PSTN Usage

 

4. Route

Control Panel > Voice Routing > Route

Figure : Edit Voice Route

 

 

5. Trunk Configuration

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 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 : Telus

 
 

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 are sequentially selected from there. In addition, Transformation tables are configurable as a reusable pool that Action Sets can reference.

Figure : From Telus

 

Figure : From Skype

 

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 Telus

 

Figure : From Skype to Telus

 
 

7. Condition Rule Tables

Select Settings > Message Manipulation > Condition Rule Tables.

Condition rules are simple rules that apply to a specific component of a message (for example, diversion.uri.host, from.uri.host, etc.) the value of the field specified in the Match Type list box can be matched against a literal value, token, or REGEX.

Figure : Conditions Rule Tables

 

8. Message Rule Tables

Select Settings > Message Manipulation > Message Rule Tables

Message Rule Tables are simply sets of Condition Rules and are applied in SIP Signaling Groups when Message Manipulation is enabled.

Figure : Telus Outbound

 

 

 

9. 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
Basic inbound/outbound call
TELUS_TC1.1Call from PSTN phone to IP PBX phone
1.  Confirm 2-way voice
2. Confirm the proper calling number is shown
3. Confirm the proper call display name is shown 
 Pass 
TELUS_TC1.2Call from IP PBX phone to PSTN phone
1.  Confirm 2-way voice
2. Confirm the proper calling number is shown
3. Confirm the proper call display name is shown 
 Pass 
Call Transfer (Blind transfer)
TELUS_TC1.9PSTN phone calls IP PBX phone 1
IP PBX phone 1 performs a blind transfer to another PSTN
Confirm both way audio. 
 Pass 
TELUS_TC1.10PSTN phone calls IP PBX phone 1
IP PBX phone 1 performs a blind transfer to another PSTN
Confirm both way audio.
Repeat the same test using SIP REFER
 Pass 
Call Transfer (Consult transfer)
TELUS_TC1.13PSTN phone calls IP PBX phone 1
IP PBX phone 1 performs a consult transfer to another PSTN
Confirm both way audio. 
 Pass 
TELUS_TC1.14PSTN phone calls IP PBX phone 1
IP PBX phone 1 performs a consult transfer to another PSTN
Confirm both way audio.
Repeat the same test using SIP REFER
 Pass 
Call Forwarding Unconditional
TELUS_TC1.16Configure IP PBX phone 1 to CFU to PSTN phone
from PSTN calls phone 1 and should CFU to PSTN phone
1. Confirm 2-way voice
2. Confirm phone 1 number and display at PSTN phone
 Pass 
Call Forwarding Busy
TELUS_TC1.17Configure IP PBX phone 1 to CFB to PSTN phone
IP PBX phone 2 calls phone 1 and should CFB toPSTN phone
1. Confirm 2-way voice
2. Confirm phone 1 number and display at PSTN phone
 Pass 
Call Forwarding Don’t Answer
TELUS_TC1.20Configure IP PBX phone 1 to CFDA to PSTN phone
from PSTN calls phone 1 and should CFDA to PSTN phone
1. Confirm 2-way voice
2. Confirm phone 1 number and display at PSTN phone
 Pass 
Voicemail
TELUS_TC1.21IP PBX phone 1 calls PSTN phone,
Don't answer the call in the PSTN phone; after 4 ring, voicemail kick in
Record a message
Follow the prompt to play back the message
Follow the prompt to cancel the recording then hang up. 
 Pass 
Conference call
TELUS_TC1.23PSTN phone calls IP PBX phone 1
IP PBX phone 1 performs a conference call with PSTN
Confirm audio with PSTN phone and IP PBX phone
 Pass 
DTMF
TELUS_TC1.24

Test Inband DTMF by programming PBX end point:
From PBX dial 647-837-0597 ( conference bridge)
When hearing the prompt, enter valid Telus conference code 3369709. Follow prompts and verify connected to conference bridge.
Verify that pressed keys are recognized and successfully accessed conference bridge.
Verify by calling to conference bridge from PSTN.

 Pass 
TELUS_TC1.25

Test RFC2833 by programming PBX endpoint:
From PBX dial 647-837-0597 ( conference bridge)
When hearing the prompt, enter valid Telus conference code 3369709.
Verify that pressed keys are recognized and successfully accessed conference bridge.
Verify by calling to conference bridge from PSTN.

 Pass 
Long calls - minimum recommendation
TELUS_TC1.28long duration call on hold: Call to PSTN, PBX places call on hold for 10 min, resume call, verify 2 way audio Pass 
FAX
TELUS_TC1.29Repeat the test by setup the call with G.711. Outbound (from IP PBX to PSTN) T.38 testing , set up the call with G711, PBX re-invite with T38. verified the fax passed with T.38. Pass 
TELUS_TC1.30Repeat the test by setup the call with G.729. Outbound (from IP PBX to PSTN) T.38 testing , set up the call with G729, PBX re-invite with T38. verified the fax passed with T.38. Pass 
TELUS_TC1.31Setup the call with  G.711 as preferred codec . Outbound (from IP PBX to PSTN) FAX G.711 pass-through testing,test G711 fax pass through. Pass 
TELUS_TC1.32Inbound (from PSTN to IP PBX) T.38 testing Pass 
TELUS_TC1.33Inbound (from PSTN to IP PBX) FAX G.711 pass-through testing Pass 
International and Caribbean   
TELUS_TC1.34

PBX call International number 011442070046000
Verify two-way audio and DTMF

 Pass 
TELUS_TC1.35

PBX call Caribbean number 12463672300
Verify two-way audio and DTMF

 Pass 

*REFER method is being handled by the SBC and not being passed to provider.

 

Conclusion

These Application Notes describe the configuration steps required for the 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.