This configuration guide is for Sonus SBC 1000/2000 series (Session Border Controller) when connecting to Skype for Business 2015 and TELUS SIP Trunking using IP Authentication.
This configuration guide supports features described on the Microsoft Technet https://technet.microsoft.com/ web site.
The interoperability compliance testing focuses on verifying inbound and outbound calls flows between Sonus SBC 1000/2000 and Skype for Business 2015.
This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. 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.
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 equipment and software were used for the sample configuration provided:
Requirements
Equipment | Software Version | |
---|---|---|
Sonus Networks | SBC 2000 | V6.0.1build441 |
Tenor AFM200 | P108-09-26 | |
Applicable SBC versions | V6.0.XbuildXXX | |
Third-party Equipment
| Microsoft Skype for Business 2015 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 2000
For any questions regarding this document or the content herein, please contact your maintenance and support provider.
The testing was executed with the Telus test plan. The following features were tested for PSTN, BVOIP, Mobile, VoLTE and HGHP clients:
SIP Calls
The following new configurations are included in this section:
Topology Builder > Shared Components > PSTN Gateways
Define a new IP/PSTN Gateway
Define FQDN
Define IP Address
Define Root Trunk
Control Panel > Voice Routing > Voice Policy
Edit Voice Policy
Control Panel > Voice Routing > PSTN Usage
View PSTN Usage
Control Panel > Voice Routing > Route
Edit Voice Route
Control Panel > Voice Routing > Trunk Configuration
Edit Trunk Configuration
The following steps provide an example of how to configure Sonus SBC 1000/2000:
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
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
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
Select Settings > Media > Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
Media List
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
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
From Skype to Telus
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
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
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
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 | ||
Call Transfer (Blind transfer) | ||||
TELUS_TC1.9 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a blind transfer to another PSTN Confirm both way audio. | Pass | ||
TELUS_TC1.10 | 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 | PSTN phone calls IP PBX phone 1 IP PBX phone 1 performs a consult transfer to another PSTN Confirm both way audio. | Pass | ||
TELUS_TC1.14 | 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_TC1.16 | 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_TC1.17 | 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 | ||
Call Forwarding Don’t Answer | ||||
TELUS_TC1.20 | 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_TC1.21 | 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_TC1.23 | 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_TC1.24 | Test Inband DTMF by programming PBX end point: | Pass | ||
TELUS_TC1.25 | Test RFC2833 by programming PBX endpoint: | Pass | ||
Long calls - minimum recommendation | ||||
TELUS_TC1.28 | 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_TC1.29 | 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_TC1.30 | 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_TC1.31 | 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_TC1.32 | Inbound (from PSTN to IP PBX) T.38 testing | Pass | ||
TELUS_TC1.33 | Inbound (from PSTN to IP PBX) FAX G.711 pass-through testing | Pass | ||
International and Caribbean | ||||
TELUS_TC1.34 | PBX call International number 011442070046000 | Pass | ||
TELUS_TC1.35 | PBX call Caribbean number 12463672300 | Pass |
*REFER method is being handled by the SBC and not being passed to provider.
These Application Notes describe the configuration steps required for the 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.