Table of Contents
This Application Note is a configuration guide for the Sonus SBC 1000/2000 Series (Session Border Controller) when connecting to Skype for Business 2015 (Skype 2015) and AT&T IP Flexible Reach SIP Trunk.
This configuration guide supports features provided in the Microsoft Technet web page:
- For additional information on Skype 2015, please visit http://microsoft.com
- For additional information on Sonus SBC 1000/2000, please visit http://sonus.net.
The interoperability compliance testing focuses on verifying inbound and outbound call flows between Sonus SBC 1000/2000, Skype 2015 and AT&T IP Flexible Reach SIP Trunk.
This is a technical document intended for telecommunications engineers for the purpose of configuring both the Sonus SBC and third-party vendor hardware. There are steps that require navigating third-party hardware interfaces as well as the Sonus SBC Command Line (CLI) interface. Understanding the basic concepts of TCP/UDP, IP routing, and SIP/RTP are required 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.
The following table lists the software and hardware equipment used in the provided test configuration.
Equipment Software Version
Vendor Sonus Networks SBC 1000 V5.0.3build407 Tenor AF P108-09-21 Third-party Microsoft Microsoft Skype for Business 2015 (Skype 2015) Mediation Server 6.0.9319.0 Polycom Polycom CX600 SIP Phone 4.0.7577.44455
The following reference configuration topology shows connectivity between third-party equipment and the Sonus SBC 1000/2000.
For any questions regarding this document or the content herein, please contact your maintenance and support provider.
Third-party Product Features
The following third-party product features are supported:
- Basic originated and terminated calls
- Basic inbound/outbound call
- Hold and Resume
- Call Forwarding
- Conference Call
Not Supported Features
- Network-Based Blind Call Transfer with REFER
- Network-Based Consultative Call Transfer with REFER (Attended)
- Network-Based Consultative Call Transfer with REFER (Unattended)
No special licensing is required for this test.
Skype 2015 Configuration
The following configuration steps are provided to configure Skype 2015 to interoperate with the Sonus SBC 1000/2000:
1. PSTN Gateway
Configure the PSTN Gateway using the following configuration screens:
2. Voice Policy
Select Control Panel > Voice Routing > Voice Policy to access the Voice Policy configuration screen.
3. PSTN Usage
Select Control Panel > Voice Routing > PSTN Usage to access the PSTN Usage configuration screen.
Select Control Panel > Voice Routing >Route to access the Route configuration screen.
5. Trunk Configuration
Select Control Panel > Voice Routing >Trunk Configuration to access the trunk configuration screen.
Sonus SBC 1000/2000 Configuration
The following configuration steps provide an example of how to configure the Sonus SBC 1000/2000 to interoperate with Skype 2015 and AT&T IP Flexible Reach SIP Trunk:
1. SIP Profile
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.
Select Settings > SIP > SIP Profiles to access the SIP Profile screen.
The default SIP profile used for the SBC 1000/2000 for this testing effort is provided in the following figures.
2. SIP Server
SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000.
Select Settings > SIP > SIP Server Tables to access the SIP Server Tables screen.
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, as shown in the following figures.
3. 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.
Select Settings > Media > Media Profiles.
Shown in the following figures are the media profiles of the voice codecs used for the SBC 1000/2000 in this testing effort and are provide for reference only.
4. Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
Select Settings > Media > Media List to access the Media List configuration screen.
5. 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.
Select Settings > Signaling Groups to access the Signaling Groups configuration screens.
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 will be configurable as a reusable pool that Action Sets can reference.
Select Settings > Transformation to access the Transformation configuration screen.
7. 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).
Select Settings > Call Routing Table to access the Call Routing Table configuration screen.
8. Cause Call Reroute
Terminating any calls return a Q.850 Cause Code when they end. We can use these codes to determine whether or not to reroute the call to another signalling group. A Cause Code Reroute table contains one or more Q.850 Cause Codes which, when matched, trigger a reroute.
Select Settings > Telephony Mapping Tables > Cause Code Reroute to access the Cause Code Reroute configuration screen.
9. 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.
This Application Note describes the configuration steps required for the Sonus SBC 1000/2000 to successfully interoperate with Skype for Business 2015 server and AT&T IP Flexible Reach SIP Trunk. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Not Supported Section.