Table of Contents
Document Overview
This document provides a configuration guide for Sonus SBC 1000/2000 series (Session Border Controller) when connecting to Skype for Business 2015 and TELUS SIP Trunking using SIP Registration.
This configuration guide supports features given in Microsoft Technet web page.
- For additional information on Skype for Business 2015, please visit http://microsoft.com
- 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 Sonus SBC 1000/2000 and Skype for Business 2015.
Audience
This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus SBC 1000/2000 aspects of the TELUS IP Trunking 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:
Requirements
Equipment | Software Version | |
|---|---|---|
| Sonus Networks | SBC 1000 | V5.0.5build418 |
| Tenor AFM200 | P108-09-26 | |
| Applicable SBC versions | V5.0.XbuildXXX | |
Third-party Equipment
| Microsoft Skype for Business 2015 (SFB2015) Mediation Server | 6.0.9319.0 |
| Polycom CX600 SIP Phone | 4.0.7577.44455 | |
VentaFax | 7.6.243.597 I |
Reference Configuration
The following reference configuration shows connectivity between third-party and Sonus SBC 1000/2000.
Connectivity Between Third-Party and Sonus SBC 1000
Support
Technical support for Sonus SBC 1000/2000 is available via phone or logging a trouble ticket.
- Phone: (978) 614-8589 or (888) 391-3434 (Toll-free)
- Web: http://www.sonus.net/company/maintenance/log-trouble-tickets
Third-Party Product Features
The testing was executed with the Telus test plan. The following features were tested for PSTN, BVOIP, Mobile, VoLTE and HGHP clients:
- Basic originated and terminated calls
- Basic inbound/outbound call
- Basic inbound/outbound call with privacy
- Hold and resume
- Call Transfer (Blind transfer)
- Call Transfer (Consult transfer)
- Call Forwarding Unconditional
- Call Forwarding Busy
- Call Forwarding Don’t Answer
- Voicemail
- Conference call
- Long calls
- FAX
- DTMF
Verify License
SIP, DSP.
Skype for Business 2015 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 1000/2000 Configuration
The following steps provide an example of how to configure 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
Telus
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 are sequentially selected from there. In addition, Transformation tables are configurable as a reusable pool that Action Sets can reference.
From Telus
From Skype
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 Telus to Skype
From Skype to Telus
7. Condition Rule Tables
Select Settings > Message Manipulation > Condition Rule Tables.
Condition rules are simple rules that apply to a specific component of a message (for example, diversion.uri.host, from.uri.host, etc.) the value of the field specified in the Match Type list box can be matched against a literal value, token, or REGEX.
Conditions Rule Tables
8. Message Rule Tables
Select Settings > Message Manipulation > Message Rule Tables
Message Rule Tables are simply sets of Condition Rules and are applied in SIP Signaling Groups when Message Manipulation is enabled.
Telus Outbound
9. Contact Registrant Table
Select Settings > Contact Registrant Table
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 can be used for outbound calls.
Telus Contact Registration Table
10. Remote Authorization Tables
Select Settings > 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).
Telus Remote Authorization Table
11. 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 |
|---|---|---|---|---|
| Basic inbound/outbound call | ||||
| TELUS_TC1.1 | Call from PSTN phone to IP PBX phone 1. Confirm 2-way voice 2. Confirm the proper calling number is shown 3. Confirm the proper call display name is shown | Pass | ||
| TELUS_TC1.2 | Call from IP PBX phone to PSTN phone 1. Confirm 2-way voice 2. Confirm the proper calling number is shown 3. Confirm the proper call display name is shown | Pass | ||
| Basic inbound/outbound call with privacy | ||||
| TELUS_TC1.3 | Call from PSTN phone to IP PBX phone, prefix the IP PBX phone number with *63 1. Confirm 2-way voice 2. Confirm the proper calling number is not shown 3. Confirm the proper call display name is not shown | Pass | ||
| TELUS_TC1.4 | Call from IP PBX phone to PSTN phone, when dialling from the IP PBX phone, use the prefix if applicable to temporary suppress the call display 1. Confirm 2-way voice 2. Confirm the proper calling number is not shown 3. Confirm the proper call display name is not shown | Pass | ||
| Hold and resume | ||||
| TELUS_TC1.5 | Call from PSTN to IP PBX - after the call setup the PBX phone puts the call on-hold or (MOH), waits 30 seconds, resumes. Confirm audio both way after resume. | Pass | ||
| TELUS_TC1.6 | Call from IP PBX to PSTN - after the call setup, use PSTN phone to put the call on-hold, wait 30 seconds, resume. Confirm audio both way after resume. | Pass | ||
| Call Transfer (Blind transfer) | ||||
| TELUS_TC1.7 | IP PBX phone 1 calls IP PBX phone 2 IP PBX phone 2 performs a blind transfer to PSTN phone Confirm audio both way after the transfer | Pass | ||
| TELUS_TC1.8 | IP PBX phone 1 calls IP PBX phone 2 IP PBX phone 2 performs a blind transfer to PSTN phone Confirm audio both way after the transfer Repeat the same test using SIP REFER | Pass | ||
| TELUS_TC1.9 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a blind transfer to IP PBX phone 2 Confirm audio both way after the transfer | Pass | ||
| TELUS_TC1.10 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a blind transfer to IP PBX phone 2 Confirm audio both way after the transfer Repeat the same test using SIP REFER | Pass | ||
| TELUS_TC1.11 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a blind transfer to another PSTN Confirm both way audio. | Pass | ||
| TELUS_TC1.12 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a blind transfer to another PSTN Confirm both way audio. Repeat the same test using SIP REFER | Pass | ||
| Call Transfer (Consult transfer) | ||||
| TELUS_TC1.13 | IP PBX phone 1 calls IP PBX phone 2 IP PBX phone 2 performs a consult transfer to PSTN phone Confirm audio both way after the transfer | Pass | ||
| TELUS_TC1.14 | IP PBX phone 1 calls IP PBX phone 2 IP PBX phone 2 performs a consult transfer to PSTN phone Confirm audio both way after the transfer Repeat the same test using SIP REFER | Pass | ||
| TELUS_TC1.15 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a consult transfer to IP PBX phone 2 Confirm audio both way after the transfer | Pass | ||
| TELUS_TC1.16 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a consult transfer to IP PBX phone 2 Confirm audio both way after the transfer Repeat the same test using SIP REFER | Pass | ||
| TELUS_TC1.17 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a consult transfer to another PSTN Confirm both way audio. | Pass | ||
| TELUS_TC1.18 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a consult transfer to another PSTN Confirm both way audio. Repeat the same test using SIP REFER | Pass | ||
| Call Forwarding Unconditional | ||||
| TELUS_TC1.19 | Configure IP PBX phone 1 to CFU to PSTN phone IP PBX phone 2 calls phone 1 and should CFU to PSTN phone 1. Confirm 2-way voice 2. Confirm phone 1 number and display at PSTN phone | Pass | ||
| TELUS_TC1.20 | Configure IP PBX phone 1 to CFU to PSTN phone from PSTN calls phone 1 and should CFU to PSTN phone 1. Confirm 2-way voice 2. Confirm phone 1 number and display at PSTN phone | Pass | ||
| Call Forwarding Busy | ||||
| TELUS_TC1.21 | Configure IP PBX phone 1 to CFB to PSTN phone IP PBX phone 2 calls phone 1 and should CFB toPSTN phone 1. Confirm 2-way voice 2. Confirm phone 1 number and display at PSTN phone | Pass | ||
| TELUS_TC1.22 | Configure IP PBX phone 1 to CFB to PSTN phone from PSTN calls phone 1 and should CFB to PSTN phone 1. Confirm 2-way voice 2. Confirm phone 1 number and display at PSTN phone | Pass | ||
| Call Forwarding Don’t Answer | ||||
| TELUS_TC1.23 | Configure IP PBX phone 1 to CFDA to PSTN phone IP PBX phone 2 calls phone 1 and should CFDA toPSTN phone 1. Confirm 2-way voice 2. Confirm phone 1 number and display at PSTN phone | Pass | ||
| TELUS_TC1.24 | Configure IP PBX phone 1 to CFDA to PSTN phone from PSTN calls phone 1 and should CFDA to PSTN phone 1. Confirm 2-way voice 2. Confirm phone 1 number and display at PSTN phone | Pass | ||
| Voicemail | ||||
| TELUS_TC1.25 | IP PBX phone 1 calls PSTN phone, Don't answer the call in the PSTN phone; after 4 ring, voicemail kick in Record a message Follow the prompt to play back the message Follow the prompt to cancel the recording then hang up. | Pass | ||
| Conference call | ||||
| TELUS_TC1.26 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a conference call with IP PBX phone 2 Confirm audio among the parties | Pass | ||
| TELUS_TC1.27 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a conference call with PSTN Confirm audio with PSTN phone and IP PBX phone | Pass | ||
| DTMF | ||||
| TELUS_TC1.28 | From PBX dial 4036929600 ( conference bridge) When hearing the prompt, enter valid Telus conference code 6913642. Follow prompts and verify connected to conference bridge. Verify that pressed keys are recognized and successfully accessed conference bridge. Verify by calling to conference bridge from PSTN. Test Inband DTMF by programming PBX end point | Pass | ||
| TELUS_TC1.29 | From PBX dial 4036929600 ( conference bridge) When hearing the prompt, enter valid Telus conference code 6913642. Verify that pressed keys are recognized and successfully accessed conference bridge. Verify by calling to conference bridge from PSTN. Test RFC2833 by programming PBX endpoint | Pass | ||
| Automatic CLID Blocking | ||||
| TELUS_TC1.30 | Automatic Blocking Feature to be setup for the SIP PBX in the switch. Call from SIP PBX to a PSTN Number. 1. Confirm 2-way voice 2. Confirm the proper calling number (IPTR2 DID or Alternate Number from SIP PBX) is not shown 3. Confirm that SIP PBX is not sending out Name in the call. | N/A | ||
| Long calls - minimum recommendation | ||||
| TELUS_TC1.31 | long duration call: 10 mins - to PSTN phone | Pass | ||
| TELUS_TC1.32 | long duration call on hold: Call to PSTN, PBX places call on hold for 10 min, resume call, verify 2 way audio | Pass | ||
| FAX | ||||
| TELUS_TC1.33 | Repeat the test by setup the call with G.711. Outbound (from IP PBX to PSTN) T.38 testing , set up the call with G711, PBX re-invite with T38. verified the fax passed with T.38. | Pass | ||
| TELUS_TC1.34 | Repeat the test by setup the call with G.729. Outbound (from IP PBX to PSTN) T.38 testing , set up the call with G729, PBX re-invite with T38. verified the fax passed with T.38. | Pass | ||
| TELUS_TC1.35 | Setup the call with G.711 as preferred codec . Outbound (from IP PBX to PSTN) FAX G.711 pass-through testing,test G711 fax pass through. | Pass | ||
| TELUS_TC1.36 | Inbound (from PSTN to IP PBX) T.38 testing | Pass | ||
| TELUS_TC1.37 | Inbound (from PSTN to IP PBX) FAX G.711 pass-through testing | 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