This Application Note is a configuration guide for the Ribbon SBC (Session Border Controller) 2000 when connecting to Skype for Business 2015 (Skype 2015) with Swisscom Enterprise SIP Trunk. The configuration guide supports features provided on the Microsoft Technet web page.
The interoperability compliance testing focuses on verifying various inbound and outbound call flows between the Ribbon SBC 2000 and Skype 2015.
This technical document is intended for telecommunication engineers with the purpose of configuring the Ribbon SBC 2000 aspects of the Swisscom Enterprise SIP trunk group together with Skype 2015. Some steps require navigating third-party equipment and Ribbon SBC Web browser user interfaces. Understanding IP/Routing and SIP/RTP basic concepts is also necessary to complete the configuration and perform any troubleshooting.
The following table lists the equipment and software used for the provided reference configuration.
Requirements
Equipment | Version | |
---|---|---|
Ribbon Networks | SBC 2000 | V8.0.2 |
Third-Party Equipment | Microsoft Skype for Business 2015 (Skype 2015) Mediation Server | 6.0.9319.0 |
Fax Machine VentaFax | 7.6.243.616 |
The following reference configuration shows connectivity between third-party equipment and the Ribbon SBC 2000.
Connectivity Between Third-Party and Sonus SBC 2000
For any questions regarding this document or the content herein, please contact your maintenance and support provider.
The following features were tested using the SwissCom test plan:
Attended call transfer
Blind call transfer
Calling line indication presentation (CLIP)
Fax with T.38
No special licensing required.
The following new configurations are included in this section:
To configure the PSTN Gateway, select Topology Builder > Shared Components > PSTN Gateways, as shown in the following figures.
Define a new IP/PSTN Gateway
Define FQDN
Define IP Address
Define Root Trunk
To configure Voice Policy, select Control Panel > Voice Routing > Voice Policy, as shown in the following figure.
Edit Voice Policy
To configure the PSTN Usage, select Control Panel > Voice Routing > PSTN Usage, as shown in the following figure.
View PSTN Usage
To configure Route, select Control Panel > Voice Routing > Route, as shown in the following figures.
Edit Voice Route
Edit Voice Route 2
To configure the Trunk, select Control Panel > Voice Routing > Trunk Configuration, as shown in the following figure.
Edit Trunk Configuration
To configure the Dial Plan, select Control Panel > Voice Routing > Dial Plan > Normalization rules, as shown in the following figure.
Dial Plan Configuration
The following steps provide an example of how to configure Ribbon SBC 2000.
SIP Profiles control how the Ribbon SBC 2000 communicates with SIP devices. They 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.
Swisscom SIP Profile
Skype 2015 SIP Profile
Fax SIP Profile
SIP Server tables contain information about the SIP devices connected to the Ribbon 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.
Swisscom SIP Server
Skype 2015 SIP Server
Fax SIP Server
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.
Swisscom Voice Codec Media Profiles
Skype 2015 Voice Codec Media Profiles
Fax Voice Codec Media Profiles
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.
Swisscom Media Lists
Skype 2015 Media Lists
Fax Media Lists
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.
Condition Rule ID 2
Condition Rule ID 3
Condition Rule ID 5
Skype 2015 Inbound Rule - Add PAID Header
Skype 2015 Inbound Rule - Add Privacy Header
Skype 2015 Inbound Rule - PAID Host Part
Swisscom Outbound Rule - Anonymous From Header
Swisscom Outbound Rule - Anonymous Contact Header
Swisscom Outbound Rule - PAID Header from Diversion Header
For CLIR, Swisscom asked us to remove the second PAI and add in the first PAI with the calling number.
Swisscom Outbound Rule - ModifyPAID
Swisscom Outbound Rule - Delete Second PAI
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.
Swisscom Signaling Group
Skype 2015 Signaling Group
Fax Signaling Group
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.
Transformation Tables Example
Fax Tables Example
Call Routing allows calls to be carried between Signaling Groups, therefore 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.
Swisscom Call Routing Table
Skype 2015 Call Routing Table
Fax Call Routing Table
Test Results
Test Case ID | Test Case | TC Description | Notes | Result |
---|---|---|---|---|
1 | General | |||
1.1 | Keepalive | |||
1.1.5 | Keep alive out of Session from PBX to eSBC | SIP options sent every 10 seconds | Pass | |
1.1.6 | Keep alive out of Session from eSBC to PBX | SIP options sent every 10 seconds | Pass | |
2.1 | Outgoing Call Scenarios | |||
2.1.2 | PBX TDM 0xx xxxxxx --> Call Matrix | |||
2.1.4 | PBX 0049xxxxxxx--> International | Pass | ||
2.1.5 | PBX --> IN Dienste (0800 800 103) IVR selection after connect | DTMF sent as rtp event 101 | Pass | |
2.1.6 | PBX --> IN Dienste (0900 55 00 98) IVR selection before connect | DTMF sent as rtp event 101 | Pass | |
2.1.7 | Long duration call | PBX --> tdm | at least 30 minutes (two sip refreshes) | Pass | |
2.1.7 | Long duration call | PBX --> bcsh | at least 30 minutes (two sip refreshes) | Pass | |
2.1.8 | Call to UFIN number (+800 0000 0141) | Pass | ||
2.1.9 | PBX --> tdm > B party does not answer (cancel) | Pass | ||
2.2 | Incoming Call scenarios | |||
2.2.2 | tdm ---> PBX DTMF | DTMF sent as rtp event 101 | Pass | |
2.2.3 | Long duration call | 1lv --> PBX | at least 30 minutes | Pass | |
2.2.3 | Long duration call | Bcsh --> PBX | at least 30 minutes | Pass | |
2.2.4 | International --> PBX | Pass | ||
2.2.5 | TDM --> PBX > B party does not answer (cancel) | A party cancel the calls after B starts ringing | Pass | |
2.3 | Call reject | B Party rejects the call | ||
2.3.1 | PBX --> tdm | Pass | ||
2.3.1 | PBX --> cucm | Pass | ||
2.3.1 | PBX --> SfB | Pass | ||
2.3.1 | PBX --> mbc or bcsh | Pass | ||
2.3.2 | tdm --> PBX | Pass | ||
2.4 | Call busy Subscriber | B Party is busy | ||
2.4.1 | PBX --> tdm | Pass | ||
2.4.2 | tdm --> PBX | Skype 2015 does not support it | Not Supported | |
2.5 | Announcements | |||
2.5.1 | Call unknown Number | PBX --> 099 999 99 99 | Pass | ||
2.5.2 | Call to switched off Mobile Number without voicemail (early media) | Pass | ||
3 | Call Indication | |||
3.1.2 | PBX --> Call Matrix, Call with CLIR | Privacy set to id, from and contact to anonymous, user number in PAI header | ||
3.1.2 | PBX --> tdm, Call with CLIR | Pass | ||
3.1.3 | TDM(*31) --> PBX | Pass | ||
4 | Tones, Announcements and ResponseCode | |||
4.1 | MOH, Announcements, ACR | |||
4.1.1 | Call Matrix --> PBX > PBX Hold > PBX Resume | MoH | ||
4.1.1 | cucm --> PBX > PBX Hold > PBX Resume | Pass | ||
4.1.1 | SfB --> PBX > PBX Hold > PBX Resume | Pass | ||
4.1.1 | mbc or bcsh --> PBX > PBX Hold > PBX Resume | Pass | ||
4.1.1 | 1lv or mykmu --> PBX > PBX Hold > PBX Resume | Pass | ||
4.1.4 | PBX --> Call Matrix > Call Matrix Hold > Call Matrix Resume | MoH | ||
4.1.4 | pbx --> tdm > tdm Hold > tdm Resume | Pass | ||
4.1.4 | pbx --> cucm > cucm Hold > cucm Resume | Pass | ||
4.1.4 | pbx --> SfB > SfB Hold > SfB Resume | Pass | ||
4.1.4 | pbx --> mbc or bcsh > mbc or bcsh Hold > mbc or bcsh Resume | Pass | ||
4.1.4 | pbx --> 1lv or mykmu > 1lv or mykmu Hold > 1lv or mykmu Resume | Pass | ||
5 | Short-Number | |||
5.1 | PBX --> Short Numbers 161 | Pass | ||
5.1 | PBX --> Short Numbers 1600 | Pass | ||
6 | Call Forwarding | |||
6.1 | CFU (Call Forwarding Unconditional) external | User C sees A number. A number in from, B number in Diversion header or history-info header | ||
6.1.1 | A TDM --> B PBX (*21) --> C Call Matrix | Function / CLIP | ||
6.1.1 | A TDM --> B PBX (*21) --> C TDM | Pass | ||
6.1.1 | A TDM --> B PBX (*21) --> C cucm | Pass | ||
6.1.1 | A TDM --> B PBX (*21) --> C SfB | Pass | ||
6.1.1 | A TDM --> B PBX (*21) --> C mbc or bcsh | Pass | ||
6.1.1 | A TDM --> B PBX (*21) --> C 1lv or mykmu | Pass | ||
6.2 | CFNA (Call Forwarding No Answer) external | User C sees A number. A number in from, B number in Diversion header or history-info header | ||
6.2.1 | A TDM --> B PBX (*61) --> C Call Matrix | Function / CLIP | ||
6.2.1 | A TDM --> B PBX (*61) --> C TDM | Pass | ||
6.2.1 | A TDM --> B PBX (*61) --> C cucm | Pass | ||
6.2.1 | A TDM --> B PBX (*61) --> C SfB | Pass | ||
6.2.1 | A TDM --> B PBX (*61) --> C mbc or bcsh | Pass | ||
6.2.1 | A TDM --> B PBX (*61) --> C 1lv or mykmu | Pass | ||
6.3 | PBX is CFU destination | |||
6.3.1 | A TDM -> B CUCM -> C PBX | Pass | ||
6.3.1 | A TDM -> B BCSH -> C PBX | Pass | ||
6.4 | Advanced forwarding scenarios | |||
6.4.1 | A TDM (CLIR enabled) -> B PBX CFU -> C TDM | C sees anonymous. Anonymous in from, B number in Diversion or history info header | Pass | |
8 | Extended Call Functions | |||
8.1 | Transfer (external) | REFER method not supported | ||
8.1.1 | TDM --> PBX (Call hold & Transfer with consultation) --> Call Matrix | A calls B, B calls C (A on hold), C picks up the call, B transfers, A and C in call, C hangs up | ||
8.1.1 | TDM --> PBX (Call hold & Transfer with consultation) --> tdm | Pass | ||
8.1.1 | TDM --> PBX (Call hold & Transfer with consultation) --> SfB | Pass | ||
8.1.2 | TDM --> PBX (Call hold & Blind Transfer) --> Call Matrix | A 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 | ||
8.1.2 | TDM --> PBX (Call hold & Blind Transfer) --> tdm | Pass | ||
8.1.2 | TDM --> PBX (Call hold & Blind Transfer) --> SfB | Pass | ||
8.2 | Transfer (internal) | |||
8.2.2 | Call Matrix --> PBX1 (Call hold & Blind Transfer) --> PBX 2 | A 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 | ||
8.2.2 | tdm--> PBX1 (Call hold & Blind Transfer) --> PBX 2 | Pass | ||
8.2.2 | SfB --> PBX1 (Call hold & Blind Transfer) --> PBX 2 | Pass | ||
8.3 | Call Matrix is transferee (with consultation) | |||
8.3.1 | PBX--->sfb--->transfer to TDM | Pass | ||
10 | Fax / Modem VBD Mode (V.152 Subset) | |||
10.1.1.2 | T.38 PBX > TDM 10 pages | Pass | ||
10.1.5.1 | T.38 PBX > Destination G.711 only 3 pages (Fallback) | Pass |
These Application Notes describe the configuration steps required for Ribbon SBC 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 and observations noted in Test Results.
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