Table of Contents
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.
- For additional information on Skype for Business 2015, visit http://microsoft.com
- For additional information on Sonus SBC 1000/2000, visit http://sonus.net
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:
Requirements
Equipment | Software Version | |
|---|---|---|
| Sonus Networks | SBC 2000 | V6.1.1build459 |
| Tenor AFM200 | P108-09-26 | |
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.
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
Select Topology Builder > Shared Components > PSTN Gateways
Define a new IP/PSTN Gateway
Define FQDN
Define IP Address
Define Root Trunk
2. Voice Policy
Select Control Panel > Voice Routing > Voice Policy
Edit Voice Policy
3. PSTN Usage
Select Control Panel > Voice Routing > PSTN Usage
View PSTN Usage
4. Route
Select Control Panel > Voice Routing > Route
Edit Voice Route
5. Trunk Configuration
Select Control Panel > Voice Routing > Trunk Configuration
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:
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.
Skype
Fax
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:
Voice Codec G711 A-Law
Voice Codec G711 U-Law
Voice Codec G729
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.
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.
From Pure IP
From Pure IP Fax
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).
From Pure IP
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.
Internal Side
External Side
Test Results
Test Results
| S.No | Procedure | Observation | Result | Comment |
|---|---|---|---|---|
| SIP Options | ||||
| 1.1.1 | Validate OPTIONS messages sent | Pass | ||
| 1.1.2 | Validate OPTIONS messages received | Pass | ||
| Initial Calls To/From PSTN | ||||
| 1.2.1 | Inbound call from a PSTN phone to an extension. Hang-up at called party. Wait for calling party to disconnect. | Pass | ||
| 1.2.2 | Inbound call from a PSTN phone to an extension. Hang-up at calling party (PSTN phone). Wait for called party to disconnect. | Pass | ||
| 1.2.3 | Outbound call from an extension to a PSTN phone. Hang-up at called party (PSTN phone). Wait for calling party to disconnect. | Pass | ||
| 1.2.4 | Outbound 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 | ||||
| 1.4.1 | Inbound call from a PSTN phone to an extension. Hang-up before far-end answers. | Pass | ||
| 1.4.2 | Outbound call from an extension to a PSTN phone. Hang-up before far-end answers. | Pass | ||
| 1.4.3 | No Answer of inbound call from a PSTN phone to an extension. | Pass | ||
| 1.4.4 | No Answer of outbound call from an extension to a PSTN phone | Pass | ||
| 1.4.5 | Inbound call from a PSTN phone to an extension that is “Busy”. | N/A | Skype for Business doesn’t send 486 Busy | |
| 1.4.6 | Outbound call from an extension to a PSTN phone that is “Busy”. | Pass | ||
| 1.4.7 | Outbound call from an extension to an invalid PSTN number. | Pass | ||
| DTMF Tone Support | ||||
| 1.5.1 | Outbound call to PSTN number that requires DTMF input | Pass | ||
| 1.5.2 | Inbound call to number that requires DTMF input | Pass | ||
| PSTN Numbering Plans | ||||
| 1.6.1 | Outbound Toll-Free Call | Pass | ||
| 1.6.2 | Outbound Local Call | Pass | ||
| 1.6.3 | Outbound International Calls | Pass | ||
| 1.6.4 | Local Directory Assistance Call or special number | Pass | ||
| 1.6.5 | Emergency Calls | N/A | Upon agreement this test is skipped not to cause any issues with local PSP. | |
| Calling Name / Number Blocking | ||||
| 1.7.1 | Inbound Calling Party Number (CPN) Block from PSTN phone | Pass | ||
| 1.7.2 | Outbound Calling Party Number (CPN) Block from phone | Pass | ||
| Supplementary Features – Call Hold, Forward, Transfer, Conference | ||||
| 1.8.1 | Inbound Call – PBX Hold and Resume – Short Duration | Pass | ||
| 1.8.2 | Outbound Call – PBX Hold and Resume – Short Duration | Pass | ||
| 1.8.3 | Inbound Call – PSTN Hold and Resume – Short Duration | Pass | ||
| 1.8.4 | Outbound Call – PSTN Hold and Resume – Short Duration | Pass | ||
| 1.8.5 | Call - PBX Hold and Resume – Long Duration that exceeds the SIP session timers (~10 min) | Pass | ||
| 1.8.6 | Call Forward – All of inbound call from PSTN to another extension | Pass | ||
| 1.8.7 | Call Forwarding – Busy / Don’t Answer of inbound call from PSTN to another extension | Pass | ||
| 1.8.8 | Call Forwarding Off Net over SIP Trunk - Inbound call from PSTN to extension is forwarded to another PSTN endpoint. | Pass | ||
| 1.8.9 | Blind Call Transfer (xfer before answer) of inbound PSTN call transferring to internal extension | Pass | ||
| 1.8.10 | Blind Call Transfer (xfer before answer) of inbound PSTN call transferring to another PSTN endpoint. | Pass | ||
| 1.8.11 | Consultative Call Transfer (xfer after answer) of inbound PSTN call transferring to internal extension | Pass | ||
| 1.8.12 | Consultative Call Transfer (xfer after answer) of inbound PSTN call transferring to another PSTN endpoint | Pass | ||
| 1.8.13 | Consultative Call Transfer (xfer after answer) of outbound PSTN call transferring to internal extension | Pass | ||
| 1.8.14 | Conference of inbound call | Pass | ||
| 1.8.15 | Conference 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 | ||
| 1.9.3 | Incoming fax using G711 | Pass | ||
| 1.9,4 | Outbound fax using G711 | Pass | ||
| Stability and Maintenance | ||||
| 1.10.1 | Long Duration Calls | Pass | ||
| 1.10.2 | Incoming & 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.
Overview
Content Tools