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Document Overview
This configuration guide provides instructions for Sonus SBC Edge (1000/2000) Series (Session Border Controller) when deployed in support of Microsoft® Skype for Business® 2015 Server (SFB2015). A secondary goal is to demonstrate how the SBC attaches 3rd party SIP-based non-SFB2015 clients into the SFB2015 environment, including the offer to these clients of network redundancy to overcome typical failure scenarios. In this paper, Polycom® VVX® SIP endpoints are configured to assume this non-SFB2015 endpoint role.
This configuration guide supports features identified on Microsoft Technet.
- For additional information on Skype for Business 2015, please visit https://www.skype.com/en/business/
- For additional information on Sonus SBC, please visit http://sonus.net.
Introduction
The interoperability compliance testing focuses on verifying inbound and outbound calls flows between SBC 1000, its subtended clients (SIP-based endpoints, TDM/FXx endpoints/trunks, etc.) and the SFB2015 infrastructure. While all the examples refer to the SBC 1000, please note the instructions and resulting behavior are also applicable to the SBC 2000.
Audience
This technical document is intended for telecommunications engineers with the purpose of configuring both the Sonus SBC Edge, example SBC subtended endpoints, and the Skype for Business infrastructure. There will be steps that require navigating third-party as well as the Sonus SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary.
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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:
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*Tenor GW is there for demonstration purposes. The fax machine may be directly connected to the SBC 1000's FXS port if desired. | |||||||||||||||||||||||||||||
Reference Configuration
The following reference configuration shows connectivity between Skype for Business infrastructure and Sonus SBC 1000.
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Support
For any questions regarding this document or the content herein, please contact your maintenance and support provider.
Third-Party Product Features
The following call flows are supported:
- Call From PSTN to SFB2015 client: PSTN -> SBC -> SFB2015 -> SFB2015 client
- Call from PSTN to analog phones*: PSTN -> SBC -> SFB2015 -> SBC -> analog phones*
- Calls from PSTN to fax: PSTN -> SBC -> SFB2015 -> SBC -> Tenor GW -> Fax
- Calls from fax to PSTN: Fax -> Tenor GW -> SBC -> SFB2015 -> SBC -> PSTN
- Calls from analog phones* to PSTN: *analog phone -> SBC -> SFB2015 -> SBC -> PSTN
- Calls from analog phones* to SFB2015 client: analog phones* -> SBC -> SFB2015 -> SFB2015 client
- Calls from analog phones* to analog phones*: analog phones* -> SBC -> SFB2015 -> SBC-> analog phones*
- Calls from SFB2015 client to PSTN: SFB2015 client -> SFB2015 -> SBC -> PSTN
- Calls from SFB2015 client to analog phones*: SFB2015 client -> SFB2015 -> SBC -> analog phones*
- Failover scenario (SFB2015 unavailable): Call from PSTN to analog phones*: PSTN -> SBC -> analog phones*
- Failover scenario (SFB2015 unavailable): Calls from PSTN to fax: PSTN -> SBC -> Tenor GW -> Fax
- Failover scenario (SFB2015 unavailable): Calls from fax to PSTN: Fax -> Tenor GW -> SBC -> PSTN
- Failover scenario (SFB2015 unavailable): Calls from analog phones* to PSTN: analog phones* -> SBC -> PSTN
- Failover scenario (SFB2015 unavailable): Calls from analog phones* to analog phones*: analog phones* -> SBC -> analog phones*
* Please note the analog phones are the Polycom VVX SIP-based phones listed in Table 1, as opposed to an FXS based phone. These Polycom endpoints are considered as "analog" clients from the perspective of the Skype for Business Server 2015, as documented at https://technet.microsoft.com/en-us/library/gg398314(v=ocs.14).aspx
Verify License
The following SBC 1000 licensable features are required for the documented scenarios to work as described:
- SIP Calls (minimum of 1 license)
- SIP Registrations (minimum of 1 license)
Please refer to https://doc.rbbn.com/display/UXDOC61/Viewing+Licenses for a description of licensable features, and for follow-on references regarding license acquisition and submission.
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Skype for Business 2015 Server Configuration
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Configure the PSTN Gateway using the following configuration screens:
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*192.168.10.10 is the IP address of the Logical Interface assigned to the SFB2015 Signaling Group of the SBC 1000.
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Select Control Panel > Voice Routing > Voice Policy to access the Voice Policy configuration screen.
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Select Control Panel > Voice Routing > PSTN Usage to access the PSTN Usage configuration screen.
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Select Control Panel > Voice Routing > Route to access the Route configuration screen.
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Select Control Panel > Voice Routing > Trunk Configuration to access the trunk configuration screen.
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In Skype for Business Server 2015, start the Windows Power Shell (point to the Windows Start menu, click All Programs, and then click Windows Power Shell).
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To create new instance of the Analog Device that you can manage with the Skype server, use the New-CsAnalogDevice command. The following are examples to create the Analog Phone and Fax:
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PS C:\Users\administrator.SKYPE2015> New-CsAnalogDevice -LineUri tel:+1222111333 -DisplayName "Poly1" -RegistrarPool fe.
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PS C:\Users\administrator.SKYPE2015> New-CsAnalogDevice -LineUri tel:+1222111000 -DisplayName "Fax1" -RegistrarPool fe.s
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The preceding commands will create an Analog Device with Analog Phone and Fax functions. The following list describes the parameters:
- LineUri - Phone number for the analog device. The line Uniform Resource Identifier (URI) should be specified by using the E.164 format, and be prefixed by the "TEL:" prefix.
- DisplayName - Configures the Active Directory display name of the analog device.
- RegistrarPool - Fully qualified domain name (FQDN) of the Registrar pool where the contact object should be homed.
- AnalogFax - Set to True ($True) if the analog device is a fax machine. Set to False ($False) if the device is not a fax machine.
- Gateway - IP address of the PSTN gateway to be used by the analog device.
- OU - Distinguished name of the Active Directory organizational unit (OU) where the contact object should be located.
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Sonus SBC 1000/2000 Configuration
The following steps provide an example of how to configure the Sonus SBC 1000:
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Select Settings > SIP > SIP Profiles
SIP Profiles control how the Sonus SBC Edge 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 Edge for this testing effort:
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Select Settings > Security > SIP Server Tables
SIP Server Tables contain information about the SIP devices connected to the Sonus SBC Edge. 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.
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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 Edge in this testing effort and are shown for reference only:
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Select Settings > Media > Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
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Select Settings > Transformation
Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For example, Transformation Tables 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 are configurable as a reusable pool that Action Sets can reference.
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Select Settings > Telephony Mapping Tables
Terminating ISDN calls return a Q.850 Cause Code when they end. These codes can be used 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 that when matched, triggers a reroute.
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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).
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Select Settings > SIP > Local Registrars
SIP provides a registration function that allows users to upload their current locations for use by proxy servers. Registration creates bindings in a location service for a particular domain that associates an address-of-record URI with one or more contact addresses. This registrar feature is used by the subtended Polycom VVX SIP-based endpoints.
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Select Settings > SIP > Local/Pass-through Authorization Tables
Local Pass-through Tables contain entries with information about SIP endpoints. The SBC Edge uses this information to challenge SIP request messages such as REGISTER. It is used in the SIP Signaling Group when the Challenge Request is enabled.
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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. This is 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.
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The following snapshots show the Polycom VVX endpoint configuration used to accomplish SIP and RTP-based communications to the SBC. Recall the Polycom VVX phones, while they interact with the SBC through SIP and RTP, are for the purposes of the SFB2015 infrastructure, the equivalent of analog (for example, FXS-based) endpoints.
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The following diagrams help identify the signaling and media communications path between network elements, for example call flows.
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Call to/from PSTN to SFB2015 client: PSTN <-> SBC <-> SFB2015 <-> SFB2015 client
Call to/from PSTN to "analog" phones: PSTN <-> SBC <-> SGB2015 <-> SBC <-> "analog" phones
Calls to/from PSTN to fax: PSTN <-> SBC<-> SFB2015 <-> SBC <-> Tenor GW <-> Fax
- Failover scenario (SFB2015 unavailable): Call to/from PSTN to "analog" phones: PSTN -> SBC -> "analog" phones
- Failover scenario (SFB2015 unavailable): Calls to/from PSTN to fax: PSTN -> SBC -> Tenor GW -> Fax
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| S.No | Procedure | Observation | Result | Comment |
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| 1 | Call From PSTN to SFB2015 client: PSTN -> SBC -> SFB2015 -> SFB2015 client |
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| Pass |
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| 2 | Call from PSTN to "analog" phones: PSTN -> SBC -> SFB2015 -> SBC -> "analog" phones |
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| Pass |
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| 3 | Calls from PSTN to fax: PSTN -> SBC -> SFB2015 -> SBC -> Tenor GW -> Fax |
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| Pass |
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| 4 | Calls from fax to PSTN: Fax ->Tenor GW -> SBC -> SFB2015 -> SBC -> PSTN |
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| Pass |
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| 5 | Calls from "analog" phone to PSTN: "analog" phone -> SBC -> SFB2015 -> SBC -> PSTN |
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| Pass |
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| 6 | Calls from "analog" phone to SFB2015 client: "analog" phone -> SBC -> SFB2015 -> SFB2015 client |
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| Pass |
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| 7 | Calls from "analog" phone to "analog" phone: "analog" phone -> SBC -> SFB2015 -> SBC -> "analog" phone |
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| Pass |
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| 8 | Calls from SFB2015 Client to PSTN: SFB2015 client -> SFB2015 -> SBC - >PSTN |
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| 9 | Calls from SFB2015 to "analog" phones: SFB2015 client -> SFB2015 -> SBC -> "analog" phone |
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| 10 | Failover scenario (SFB2015 unavailable): Call from PSTN to "analog" phones: PSTN -> SBC -> "analog" phones |
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| Pass | SBC 1000 assumes the role of the backup SIP server, and routes the call to the analog phone directly based on its own routing tables | ||
| 11 | Failover scenario (SFB2015 unavailable): Calls from PSTN to fax: PSTN -> SBC -> Tenor GW -> Fax |
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| Pass | SBC 1000 assumes the role of the backup SIP server, and routes the call to the Fax directly based on its own routing tables | ||
| 12 | Failover scenario (SFB2015 unavailable): Calls from fax to PSTN: Fax -> Tenor GW -> SBC -> PSTN |
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| Pass | SBC 1000 assumes the role of the backup SIP server, and routes the call to the PSTN directly based on its own routing tables | |
| 13 | Failover scenario (SFB2015 unavailable): Calls from "analog" phone to PSTN: "analog" phone -> SBC -> PSTN |
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| Pass | SBC 1000 assumes the role of the backup SIP server, and routes the call to the PSTN directly based on its own routing tables | |
| 14 | Failover scenario (SFB2015 unavailable): Calls from "analog" phone to "analog" phone: "analog" phone -> SBC -> "analog" phone |
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| Pass | SBC 1000 assumes the role of the backup SIP server, and routes the call to the analog phone directly based on its own routing tables |
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Conclusion
This Application Note describe the configuration steps required for Sonus SBC Edge to successfully interoperate with with SFB2015. The document also successfully demonstrates how the SBC provides call services to subtended clients in the the event of a network failure or service disruption regarding the SFB2015 environment. All feature and serviceability test cases were completed and passed with the exceptions and observations noted in Test Results.
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