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Document Overview

Configuration guide describes connecting the Sonus Session Border Controller (SBC) 2000 to Skype for Business 2015 (Skype 2015) with SwissCom Enterprise SIP Trunk, and supports features provided on the Microsoft Technet web page.

Introduction

The interoperability compliance testing focuses on verifying various inbound and outbound call flows between Sonus SBC 2000 and Skype 2015.

Audience

This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus SBC 2000 aspects of the SwissCom Enterprise SIP trunk group together with Skype 2015. Some steps  require navigating third-party equipment and Sonus SBC Web browser user interfaces. Understanding IP/Routing and SIP/RTP basic concepts are also necessary to complete the configuration and perform any 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 provided sample configuration:

 

Table : Requirements

Equipment Version
Sonus NetworksSBC 2000 V6.1.0build457
Tenor AF P108-09-21
Third-Party EquipmentMicrosoft Skype for Business 2015 (Skype 2015) Mediation Server 6.0.9319.0

Polycom CX600 SIP Phone

4.0.7577.44455


 

 

Reference Configuration

The following reference configuration shows connectivity between third-party equipment and Sonus SBC 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 following features were tested using the SwissCom test plan:

  • Basic originated and terminated calls
  • Basic inbound/outbound call
  • Hold and Resume
  • Call Forwarding Unconditional
  • Call forwarding no answer
  • Attended call transfer  

  • Blind call transfer

  • 3-party conference  
  • DTMF (RFC 2833)  
  • Calling line indication presentation (CLIP)

  • Calling line indication restriction (CLIR)   
  • Fax with G.711 pass-through  
  • Fax with T.38

Verify License

No special licensing required.

Skype 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
  6. Dial Plan

1. PSTN Gateway

To configure the PSTN Gateway, select Topology Builder > Shared Components > PSTN Gateways, as shown in the following figures.

 

Figure : Define a new IP/PSTN Gateway

 

 

Figure : Define FQDN

 

 

Figure : Define IP Address

 

 

Figure : Define Root Trunk

 

2. Voice Policy

To configure Voice Policy, select Control Panel > Voice Routing > Voice Policy, as shown in the following figure.

 

Figure : Edit Voice Policy

 

3. PSTN Usage

To configure the PSTN Usage, select Control Panel > Voice Routing > PSTN Usage, as shown in the following figure.

 

Figure : View PSTN Usage

 

4. Route

To configure Route, select Control Panel > Voice Routing > Route, as shown in the following figures.

 

Figure : Edit Voice Route

 

Figure : Edit Voice Route 2

 

5. Trunk Configuration

To configure the Trunk, select Control Panel > Voice Routing > Trunk Configuration, as shown in the following figure.

 

Figure : Edit Trunk Configuration

 

6. Dial Plan

To configure the Dial Plan, select Control Panel > Voice Routing > Dial Plan > Normalization rules, as shown in the following figure.

 

Figure : Dial Plan Configuration

 

Sonus SBC 2000 Configuration

The following steps provide an example of how to configure Sonus SBC 2000.

  1. SIP Profile
  2. SIP Server
  3. Media Profile
  4. Media List
  5. Message Manipulation
  6. Signaling Groups
  7. Transformation
  8. Call Routing Table  

1. SIP Profile

SIP Profiles control how the Sonus SBC 2000 communicates with SIP devices. These control important characteristics such as: session timers, SIP Header customization, SIP timers, MIME payloads, and option tags.

To configure the SIP Profile, select Settings > SIP > SIP Profiles.

For this test effort, the default SIP profile used for the SBC 2000 is shown in the following figures.

 

Figure : SwissCom SIP Profile

Figure : Skype 2015 SIP Profile

Figure : Fax SIP Profile

 

2. SIP Server 

SIP Server tables contain information about the SIP devices connected to the Sonus SBC 2000. The table entries 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.

To configure the SIP Server, select Settings > SIP > SIP Server Tables, as shown in the following figures.

 

Figure : SwissCom SIP Server


Figure : Skype 2015 SIP Server

 

Figure : Fax SIP Server

 

3. Media Profile

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. 

To access the Media Profile, select Settings > Media > Media Profiles.

For this test effort, the following figures show the media profiles of the voice codecs used for the SBC 2000 and are provided for reference only.

 

 

Figure : SwissCom Voice Codec Media Profiles

 

 

Figure : Skype 2015 Voice Codec Media Profiles


 

Figure : Fax Voice Codec Media Profiles


 

4. Media List

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

To access Media lists, select Settings > Media > Media List, as shown in the following figures.


Figure : SwissCom Media Lists


Figure : Skype 2015 Media Lists


Figure : Fax Media Lists


 

5. Message Manipulation

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

To configure Message Manipulation, select Settings > SIP > Message Manipulation > Condition Rule Table, as shown in the following figures.

 

Figure : Condition Rule ID 2

Figure : Condition Rule ID 3

Figure : Condition Rule ID 5

Figure : Skype 2015 Inbound Rule - Add PAID Header

Figure : Skype 2015 Inbound Rule - Add Privacy Header

 
 

 

Figure : Skype 2015 Inbound Rule - PAID Host Part

Figure : SwissCom Outbound Rule - Anonymous From Header

Figure : SwissCom Outbound Rule - Anonymous Contact Header

Figure : SwissCom Outbound Rule - PAID Header from Diversion Header

 

This rule works as a workaround for the SBC to not send multiple m-line (audio and image) for SIP Re-Invite for T.38

Figure : SwissCom Inbound Rule - Change User Agent

 

 

6. 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. Signaling Groups are also the locations 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 configure Signaling Groups, select Settings > Signaling Groups, as shown in the following figures.

 

Figure : SwissCom Signaling Group

 

Figure : Skype 2015 Signaling Group

 

 

Figure : Fax Signaling Group

 

7. Transformation

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

To configure the Transformation table, select Settings > Transformation, as shown in the figures below.

 

Figure : Transformation Tables Example

 

Figure : Fax Tables Example

Figure : Fax Tables Example

 

 

8. Call Routing Table

Call Routing allows calls to be carried between Signaling Groups, thus allowing calls to be carried between ports and between protocols (for example, ISDN to SIP). Routes are defined by Call Routing tables, which allows for flexible configuration of calls that are carried, as well as 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.

To configure the Call Routing Table, select Settings > Call Routing Table, as shown in the following figures.


 

Figure : SwissCom Call Routing Table

Figure : Skype 2015 Call Routing Table

 

 

Figure : Fax Call Routing Table

 

 

Test Results

 

Figure : Test Results

Test Case IDTest CaseTC DescriptionNotesResult
     
1General   
1.1Keepalive   
1.1.5Keep alive out of Session from PBX to eSBCSIP options sent every 10 seconds Pass
1.1.6Keep alive out of Session from eSBC to PBXSIP options sent every 10 seconds Pass
     
2Basic Calls   
2.1Outgoing call scenariosNumbering format e164, preferred codec g711a, P-asserted ID header should be present  
2.1.1PBX IP 0xx xxxxxx --> Call MatrixA is an IP phone connected to the PBXSpecify in the notes the type of device used 
2.1.1PBX IP  --> tdmNote: verify if comfort noise is supported by the pbx Pass
2.1.1PBX IP --> cucm  Pass
2.1.1PBX IP --> SfBNote: verify if comfort noise is supported by the pbx Pass
2.1.1PBX IP --> mbc or bcsh  Pass
2.1.1PBX IP --> 1lv or mykmu  Pass
2.1.2PBX TDM 0xx xxxxxx --> Call MatrixA is an analogue phone connected to the PBXSpecify in the notes the type of device used 
2.1.2PBX TDM --> tdm  Pass
2.1.2PBX TDM --> cucm  Pass
2.1.2PBX TDM --> SfB  Pass
2.1.2PBX TDM --> mbc or bcsh  Pass
2.1.2PBX TDM --> 1lv or mykmu  Pass
    Pass
2.1.3PBX 00041xx xxxxxx --> TDM1  Pass
2.1.4PBX 0049xxxxxxx--> Intenational   Pass
2.1.5PBX --> IN Dienste (0800 800 103) IVR selection after connectDTMF sent as rtp event 101 Pass
2.1.6PBX --> IN Dienste (0900 55 00 98) IVR selection before connectDTMF sent as rtp event 101 Pass
2.1.7Long duration call | PBX --> tdmat least 30 minutes (two sip refreshes) Pass
2.1.7Long duration call | PBX --> bcshat least 30 minutes (two sip refreshes) Pass
2.1.8Call to UFIN number (+800 0000 0141)  Pass
2.1.9PBX --> tdm > B party doesn't answer (cancel)  Pass
     
2.2Incoming Call scenarios   
2.2.1Call Matrix  0xx xxx xx xx ---> PBX   
2.2.1tdm ---> PBX  Pass
2.2.1cucm ---> PBX  Pass
2.2.1SfB ---> PBX  Pass
2.2.1mbc or bcsh ---> PBX  Pass
2.2.11lv or mykmu ---> PBX  Pass
     
2.2.2tdm ---> PBX DTMFDTMF sent as rtp event 101 Pass
2.2.3Long duration call | TDM  --> PBX at least 30 minutes Pass
2.2.3Long duration call | Bcsh  --> PBX at least 30 minutes Pass
2.2.4International --> PBX  Pass
2.2.5TDM -->  PBX > B party doesn't answer (cancel)A party cancel the calls after B starts  ringing Pass
     
2.3Call rejectB Party rejects the call  
2.3.1PBX --> tdm  Pass
2.3.1PBX --> cucm  Pass
2.3.1PBX --> SfB  Pass
2.3.1PBX --> mbc or bcsh  Pass
     
2.3.2tdm  --> PBX  Pass
2.3.2cucm  --> PBX  Pass
2.3.2SfB --> PBX  Pass
2.3.2mbc or bcsh --> PBX  Pass
     
2.4Call busy SubscriberB Party is busy  
2.4.1PBX --> tdm  Pass
     
2.4.2tdm  --> PBX Skype 2015 does not support itNot Supported
     
2.5Announcements   
2.5.1Call unknow Number | PBX --> 099 999 99 99  Pass
2.5.2Call to swiched off Mobile Number without voicemail (early media)  Pass
     
3Call Indication    
3.1CLIP (Calling Line Identification Presentation) / CLIR+41800xxxxxx number in From and contact, real number must be in PAI header  
3.1.1PBX --> tdm with Special Arrangement +41800800800  Pass
3.1.1PBX --> cucm with Special Arrangement +41800800800  Pass
3.1.1PBX --> SfB with Special Arrangement +41800800800  Pass
3.1.1PBX --> mbc or bcsh with Special Arrangement +41800800800  Pass
     
3.1.2PBX --> Call Matrix, Call with CLIR Privacy set to id, from and contact to anonymous, user number in PAI header  
3.1.2PBX --> tdm, Call with CLIR   Pass
3.1.2PBX --> cucm, Call with CLIR   Pass
3.1.2PBX --> SfB, Call with CLIR   Pass
3.1.2PBX --> mbc or bcsh, Call with CLIR   Pass
     
3.1.3TDM(*31) --> PBX  Pass
4Tones, Announcements and ResponseCode   
4.1MOH, Announcements, ACR   
4.1.1Call Matrix --> PBX > PBX Hold > PBX ResumeMoH  
4.1.1tdm --> PBX > PBX Hold > PBX Resume  Pass
4.1.1cucm --> PBX > PBX Hold > PBX Resume  Pass
4.1.1SfB --> PBX > PBX Hold > PBX Resume  Pass
4.1.1mbc or bcsh --> PBX > PBX Hold > PBX Resume  Pass
4.1.11lv or mykmu --> PBX > PBX Hold > PBX Resume  Pass
     
4.1.4PBX --> Call Matrix > Call Matrix Hold > Call Matrix ResumeMoH  
4.1.4pbx --> tdm > tdm Hold > tdm Resume  Pass
4.1.4pbx --> cucm > cucm Hold > cucm Resume  Pass
4.1.4pbx --> SfB > SfB Hold > SfB Resume  Pass
4.1.4pbx --> mbc or bcsh > mbc or bcsh Hold > mbc or bcsh Resume  Pass
4.1.4pbx --> 1lv or mykmu > 1lv or mykmu Hold > 1lv or mykmu Resume  Pass
     
5Short-Number   
5.1PBX --> Short Numbers 161  Pass
5.1PBX --> Short Numbers 1600  Pass
5.2E112 Emergency Location check > PBX call --> 086756Correct Address should be played in announcementSimulator was used instead of Skype server.Passed with Exception
5.3E112 Emergency Location GeoPriv  

Not Executable

     
6Call Forwarding   
6.1CFU  (Call Forwarding Unconditional) externalUser C sees A number. A number in from, B number in Diversion header or history-info header  
6.1.1A TDM --> B PBX (*21) --> C Call MatrixFunction / CLIP  
6.1.1A TDM --> B PBX (*21) --> C TDM  Pass
6.1.1A TDM --> B PBX (*21) --> C cucm  Pass
6.1.1A TDM --> B PBX (*21) --> C SfB  Pass
6.1.1A TDM --> B PBX (*21) --> C mbc or bcsh  Pass
6.1.1A TDM --> B PBX (*21) --> C 1lv or mykmu  Pass
     
6.2CFNA  (Call Forwarding No Answer) externalUser C sees A number. A number in from, B number in Diversion header or history-info header  
6.2.1A TDM --> B PBX (*61) --> C Call MatrixFunction / CLIP  
6.2.1A TDM --> B PBX (*61) --> C TDM  Pass
6.2.1A TDM --> B PBX (*61) --> C cucm  Pass
6.2.1A TDM --> B PBX (*61) --> C SfB  Pass
6.2.1A TDM --> B PBX (*61) --> C mbc or bcsh  Pass
6.2.1A TDM --> B PBX (*61) --> C 1lv or mykmu  Pass
     
6.3 PBX is CFU destination   
6.3.1A TDM -> B CUCM -> C PBX  Pass
6.3.1A TDM -> B BCSH -> C PBX  Pass
     
6.4Advanced forwarding scenarios   
6.4.1A TDM (CLIR enabled) -> B PBX CFU -> C TDMC sees anonymous. Anonymous in from, B number in Diversion or history info header Pass
     
7Trunk Capacity / SAC VGate+ (Out of scope at the moment)   
     
8Extended Call Functions   
8.1Transfer (external) REFER method not supported 
8.1.1TDM --> PBX (Call hold & Transfer with consultation) --> Call MatrixA calls B, B calls C (A on hold), C picks up the call, B transfers, A and C in call, C hangs up  
8.1.1TDM --> PBX (Call hold & Transfer with consultation) --> tdm  Pass
8.1.1TDM --> PBX (Call hold & Transfer with consultation) --> cucm  Pass
8.1.1TDM --> PBX (Call hold & Transfer with consultation) --> SfB  Pass
8.1.1TDM --> PBX (Call hold & Transfer with consultation) --> mbc or bcsh  Pass
8.1.1TDM --> PBX (Call hold & Transfer with consultation) --> 1lv or mykmu  Pass
     
8.1.2TDM --> PBX (Call hold & Blind Transfer) --> Call MatrixA calls B, B calls C (A on hold), B transfers, A hears ringback tone, C picks up the call, A and C in call, C hangs up  
8.1.2TDM --> PBX (Call hold & Blind Transfer) --> tdm  Pass
8.1.2TDM --> PBX (Call hold & Blind Transfer) --> cucm  Pass
8.1.2TDM --> PBX (Call hold & Blind Transfer) --> SfB  Pass
8.1.2TDM --> PBX (Call hold & Blind Transfer) --> mbc or bcsh  Pass
8.1.2TDM --> PBX (Call hold & Blind Transfer) --> 1lv or mykmu  Pass
     
8.1.3Call Matrix --> PBX (Call hold & Transfer with consultation) --> TDMA calls B, B calls C (A on hold), C picks up the call, B transfers, A and C in call, C hangs up  
8.1.3cucm --> PBX (Call hold & Transfer with consultation) --> TDM  Pass
8.1.3SfB --> PBX (Call hold & Transfer with consultation) --> TDM  Pass
8.1.3mbc or bcsh --> PBX (Call hold & Transfer with consultation) --> TDM  Pass
8.1.31lv or mykmu --> PBX (Call hold & Transfer with consultation) --> TDM  Pass
     
8.1.4Call Matrix --> PBX (Call hold & Blind Transfer) --> TDMA calls B, B calls C (A on hold), B transfers, A hears ringback tone, C picks up the call, A and C in call, C hangs up  
8.1.4cucm --> PBX (Call hold & Blind Transfer) --> TDM  Pass
8.1.4SfB --> PBX (Call hold & Blind Transfer) --> TDM  Pass
8.1.4mbc or bcsh --> PBX (Call hold & Blind Transfer) --> TDM  Pass
8.1.41lv or mykmu --> PBX (Call hold & Blind Transfer) --> TDM  Pass
     
8.2Transfer (internal)   
8.2.1Call Matrix --> PBX1 (Call hold & Transfer with consultation) --> PBX 2A calls B, B calls C (A on hold), C picks up the call, B transfers, A and C in call, C hangs up  
8.2.1tdm--> PBX1 (Call hold & Transfer with consultation) --> PBX 2  Pass
8.2.1cucm --> PBX1 (Call hold & Transfer with consultation) --> PBX 2   
8.2.1SfB--> PBX1 (Call hold & Transfer with consultation) --> PBX 2   
8.2.1mbc or bcsh--> PBX1 (Call hold & Transfer with consultation) --> PBX 2   
8.2.11lv or mykmu -> PBX1 (Call hold & Transfer with consultation) --> PBX 2   
     
8.2.2Call Matrix --> PBX1 (Call hold & Blind Transfer) --> PBX 2A calls B, B calls C (A on hold), B transfers, A hears ringback tone, C picks up the call, A and C in call, C hangs up  
8.2.2tdm--> PBX1 (Call hold & Blind Transfer) --> PBX 2  Pass
8.2.2cucm --> PBX1 (Call hold & Blind Transfer) --> PBX 2   
8.2.2SfB --> PBX1 (Call hold & Blind Transfer) --> PBX 2   
8.2.2mbc or bcsh --> PBX1 (Call hold & Blind Transfer) --> PBX 2   
8.2.21lv or mykmu --> PBX1 (Call hold & Blind Transfer) --> PBX 2   
     
8.3Call Matrix is transferee (with consultation)   
8.3.1PBX--> mbc or bcsh --> transfer to TDM  Pass
8.3.1PBX--->cucm--->transfer to TDM   
8.3.1PBX--->sfb--->transfer to TDM   
     
8.4Conference   
8.4.13-pty conference internal - PBX1 initiating to 1 TDM / 1 Call MatrixFunction  
8.4.13-pty conference internal - PBX1 initiating to 1 TDM / 1 TDM  Pass
8.4.13-pty conference internal - PBX1 initiating to 1 TDM / 1 cucm  Pass
8.4.13-pty conference internal - PBX1 initiating to 1 TDM / 1 SfB  Pass
8.4.13-pty conference internal - PBX1 initiating to 1 TDM / 1 mbc or bcsh  Pass
8.4.13-pty conference internal - PBX1 initiating to 1 TDM / 1 1lv or mykmu  Pass
     
8.4.23-pty conference internal - PBX1 initiating to PBX2 / 1 Call MatrixFunction  
8.4.23-pty conference internal - PBX1 initiating to PBX2 / 1 TDM  Pass
8.4.23-pty conference internal - PBX1 initiating to PBX2 / 1 cucm  Pass
8.4.23-pty conference internal - PBX1 initiating to PBX2 / 1 SfB  Pass
8.4.23-pty conference internal - PBX1 initiating to PBX2 / 1 mbc or bcsh  Pass
8.4.23-pty conference internal - PBX1 initiating to PBX2 / 1 1lv or mykmu  Pass
     
9Special PBX Function   
9.1For Example Call Deflection, Park, Mobile integration or others if not tested with Vendor own Testlist   
     
10Fax / Modem VBD Mode (V.152 Subset)   
10.1Outgoing Fax T.38   
10.1.1.1T.38 PBX > TDM 3 pages  Pass
10.1.1.2T.38 PBX > TDM 10 pages  Pass
10.1.5.1T.38 PBX > Destination G.711 only 3 pages (Fallback)  Pass
     
10.2Incoming Fax T.38   
10.2.1.1TDM > T.38 PBX 3 pages  Pass
10.2.1.2TDM > T.38 PBX 10 pages  Pass
10.2.5.1Destination G.711 only > T.38 PBX 3 pages (Fallback)  Pass
     
10.3Outgoing Fax G.711 passtrough   
10.3.1.1G.711 PBX > TDM 3 pages  Pass
10.3.1.2G.711 PBX > TDM 10 pages  Pass
     
10.4Incoming Fax G.711 passtrough   
10.4.1.1TDM > G.711 PBX 3 pages  Pass
10.4.1.2TDM > G.711 PBX 10 pages  Pass


Conclusion

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