Sonus SBC 2000 V5.0.3 IOT Skype for Business 2015 SwissCom SIP Trunk Application Notes

Document Overview

This document is a configuration guide for the Sonus SBC 2000 (Session Border Controller) when connecting to Skype for Business 2015 (Skype 2015) and SwissCom 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 2000, please visit http://sonus.net.

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 SIP trunk group together with Skype 2015. Some steps will 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.

Requirements

The following equipment and software were used for the provided sample configuration:

Reference Configuration

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

Connectivity Between Third-Party and Sonus SBC 2000

Support

Technical support for Sonus SBC 2000 series is available via phone or logging a trouble ticket:

• Phone:   (978) 614-8589  or (888) 391-3434 (Toll-free)
• Web:    http://sonusnetworks.force.com/PortalLoginPage

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
• FAX
• DTMF 

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.

Define a new IP/PSTN Gateway

Define FQDN

Define IP Address

Define Root Trunk

2. Voice Policy

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

View PSTN Usage

4. Route

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

5. Trunk Configuration

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

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 Lists
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.

SwissCom 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.

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.

SwissCom Voice Codec Media Profiles

4. Media Lists

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.

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.

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.

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.

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.

Fax Call Routing Table

Interoperability Test Results

The following table provides results from interoperability compliance testing.

Exceptions

Skype 2015 rejects SIP Re-Invite (usually hold method) from a caller side immediately after the first SIP ACK message is received (acknowledge for the first SIP Invite) with SIP 491 response. SBC follows RFC3261 and starts the random timer between 2.1 and 4 seconds after it tries to send SIP Re-Invite again. During the first 10 second time period, the Skype client is not able to put the caller on hold. The call itself is not affected as the connection is on a dual direction (both way) media. The Skype client can put the caller side on hold after this time period. This behaviour is found for some specific endpoints.