Table of Contents
Document Overview
This document provides a configuration guide for Sonus SBC 2000 (Session Border Controller) when connecting to Skype for Business 2015 (SFB2015) and Intermedia SIP Trunk.
This configuration guide supports features provided in Microsoft Technet web page.
- For additional information on SFB2015, 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 calls flows between Sonus SBC 2000 and SFB2015.
Audience
This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus SBC 2000 aspects of the Intermedia SIP trunk group together with the SFB2015. Some steps will require navigating a third-party and Sonus SBC Web browser user interface. Understanding IP/Routing and SIP/RTP basic concepts are also necessary 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.
Requirements
The following equipment and software were used for the sample configuration provided:
| Equipment | Version | |
|---|---|---|
| Sonus Networks | SBC 2000 | V5.0.1build399 |
| Third-Party Equipment | Microsoft Skype for Business 2015 (SFB2015) Mediation Server | 6.0.9319.0 |
Polycom CX600 SIP Phone | 4.0.7577.44455 |
Reference Configuration
The following reference configuration shows connectivity between third-party 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 Intermedia 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.
SFB2015 Configuration
The following new configurations are included in this section:
1. PSTN Gateway
Topology Builder > Shared Components > PSTN Gateways
Define a new IP/PSTN Gateway
Define FQDN
Define IP Address
Define Root Trunk
2. Voice Policy
Control Panel > Voice Routing > Voice Policy
Edit Voice Policy
3. PSTN Usage
Control Panel > Voice Routing > PSTN Usage
View PSTN Usage
4. Route
Control Panel > Voice Routing > Route
Edit Voice Route
5. Trunk Configuration
Control Panel > Voice Routing > Trunk Configuration
Edit Trunk Configuration
Sonus SBC 2000 Configuration
The following steps provide an example of how to configure Sonus SBC 2000.
- SIP Profile
- SIP Server
- Media Profile
- Media Lists
- Contact Registrant Tables
- Remote Authorization Tables
- Signaling Groups
- Transformation
- Call Routing Table
1. SIP Profile
Select Settings > SIP > SIP Profiles
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. The default SIP profile used for the SBC 2000 for this testing effort is shown below.
Default SIP Profile
2. SIP Server
Select Settings > SIP > SIP Server Tables
SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 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.
SIP Server
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. Listed below are the media profiles of the voice codecs used for the SBC 2000 in this testing effort and is for reference only.
Voice Codec Media Profiles
SFB uses default Media List
4. Media Lists
Select Settings > Media > Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
Media Lists
5. Contact Registrant Tables
Select Settings > SIP > Contact Registrant Tables
Contact Registrant Tables are used to manage contacts that are registered to a SIP server. The SIP Server Configuration can specify a Contact Registrant Table, and the username portion of the table will be used for outbound calls.
Contact Registrant Table Example
6. Remote Authorization Tables
Select Settings > SIP > Remote Authorization Tables
Remote Authorization Tables and their entries contain information used to respond to request message challenges by an upstream server. The Remote Authorization Tables on this page appear as options in Creating and Modifying Entries in the SIP Servers (search online SBC 1000/2000 documentation).
Remote Authorization Table Example
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.
Signaling Groups Example
8. Transformation
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 are sequentially selected from there. In addition, Transformation tables will be configurable as a reusable pool that Action Sets can reference.
Transformation Tables Example
9. 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).
Call Routing Table
Test Results
| S.No | Procedure | Observation | Result | Comment |
|---|---|---|---|---|
1.1 | Customer send REGISTER and accept 401 response for authentication | AccessLine may set the Registration expiration time to 30 seconds in order to refresh NAT router binding | Passed | |
1.2 | If customer has other means of NAT router refresh messages, such as OPTIONS, STUN etc. |
| N/A | |
2.1 | Normal Call Customer initiate call, Customer hang-up | 180/183 + Answer BYE from customer | Passed | |
2.2 | Normal Call Customer initiate call, AccessLine hang-up | 180/183 + Answer BYE from AccessLine | Passed | |
2.3 | No answer call Customer initiate call, Customer hang-up before answer | 180/183 + CANCEL | Passed | |
2.4 | Busy call Customer calls a busy number | 180/183 + 486 | Passed | |
2.5 | Not Found Customer calls a non routable number | 404/480 or any other 4xx/5xx/6xx The calling party should play fast busy to the caller. | Passed | |
2.6 | 10 min call Customer initiate call | 180/183 + Answer AccessLine SBC doesn’t use session timer, it does detect RTP timeout | Passed | |
2.7 | Fax/Modem call Customer initiate call Test with both G729 and G711 | 180/183 + Answer with voice coders If the initial codec is G729, Accessline sends a reINVITE with G711 | Passed | |
2.8 | Anonymous From Customer initiate call with From=Anonymous | 180/183 + Answer | Passed | |
2.9 | G.729 Customer initiate call with SDP=G.729 , G.711 DTMF RFC2833 in G.729 | 180/183 + Answer with G.729
2 way DTMF | Passed | |
2.10 | G.711 Customer initiate call with SDP=G.711 , G.729 DTMF RFC2833 in G.711 | 180/183 + Answer with G.711
2 way DTMF | Passed | |
2.11 | AccessLine changes codecs after answer Customer initiate call with SDP=G.729 and G.711 | 1. AccessLine sends 180/183/200 SDP=G.729 2. AccessLine sends a reINVITE with SDP=G.711
| Passed | |
2.12 | Customer changes codecs after answer Customer initiate call with SDP=G.729 and G.711 | 1. AccessLine sends 180/183/200 SDP=G.729 2. Customer sends a reINVITE with SDP=G.711 | N/A | SFB does not support it |
2.13 | RTCP | Check RTCP from both AccessLine and customer | Passed | |
3.1 | Normal Call AccessLine initiate call, AccessLine hang-up | 180/183 + Answer BYE from AccessLine | Passed | |
3.2 | Normal Call AccessLine initiate call, Customer hang-up | 180/183 + Answer BYE from Customer | Passed | |
3.3 | No answer call AccessLine initiate call, AccessLine hang-up before answer | 180/183 + CANCEL | Passed | |
3.4 | Busy call AccessLine calls a busy number | 180/183 + 486 | N/A | SFB does not support it |
3.5 | Not Found AccessLine calls a non routable number | 404 or any 4xx/5xx/6xx | Passed | |
3.6 | 10 min call AccessLine initiate call | 180/183 + Answer AccessLine SBC doesn’t use session timer, it does detect RTP timeout | Passed | |
3.7 | Fax/modem call AccessLine initiate call Test with both G729 and G711 | 180/183 + Answer with voice coders If the initial codec is G729, the customer should detect the fax/modem answering tone and send a reINVITE with G711. | Passed | |
3.8 | G.729 AccessLine initiate call with SDP=G.729 , G.711 DTMF RFC2833 in G.729 | 180/183 + Answer
2 way DTMF | Passed | |
3.9 | G.711 AccessLine initiate call with SDP=G.711 , G.729 DTMF RFC2833 in G.711 | 180/183 + Answer
2 way DTMF | Passed | |
3.10 | Anonymous From AccessLine initiate call with From=Anonymous | 180/183 + Answer | Passed | |
3.11 | AccessLine changes codec after answer AccessLine initiate call with SDP=G.729 | 1. Customer sends 180/183/200 SDP=G.729 2. AccessLine sends reinvite with SDP=G.711 | Passed | |
3.12 | Customer changes codec after answer AccessLine initiate call with SDP=G.711 | 1. Customer sends 180/183/200 SDP=G.711 2. Customer sends reinvite with SDP=G.729 | N/A | SFB does not support it |
3.13 | RTCP | Check RTCP from both AccessLine and customer | Passed | |
4.1 | Call forward on busy | There are two possible implementations:
| N/A | SFB does not support it |
4.2 | Call forward on no answer | There are two possible implementations:
| Passed | |
4.3 | Call transfer | There are two possible implementations:
| Passed | |
4.4 | Local Conference | The PBX mixes the audio from a few calls | Passed | |
4.5 | Call Reject | The PBX rejects a call with 4xx status | Passed | |
4.6 | Hold / Unhold | The PBX sends a reINVITE with hold indication | Passed |
Conclusion
This Application Notes document describes the configuration steps required for Sonus SBC 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
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