This document provides a configuration guide for Ribbon SBC Edge Series (Session Border Controller) when connecting to Skype for Business 2015 and Colt SIP Trunking.
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 Ribbon SBC Edge and Skype for Business 2015.
This is a technical document intended for telecommunications engineers with the purpose of configuring both the Ribbon SBC and the third-party product. There will be steps that require navigating third-party as well as the Ribbon 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 Ribbon 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:
|Microsoft Skype for Business 2015 Mediation Server||6.0.9319.0|
|Polycom CX600 SIP Phone|
The following reference configuration shows connectivity between third-party and Ribbon SBC Edge.
For any questions regarding this document or the content herein, please contact your maintenance and support provider.
The testing was executed with the Colt test plan. The following features were tested:
The following new configurations are included in this section:
Topology Builder > Shared Components > PSTN Gateways
Control Panel > Voice Routing > Voice Policy
Control Panel > Voice Routing > PSTN Usage
Control Panel > Voice Routing > Route
Control Panel > Voice Routing > Trunk Configuration
The following steps provide an example of how to configure Ribbon SBC 1000/2000:
Select Settings > SIP > SIP Profiles
SIP Profiles control how the Ribbon 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 figures shows the default SIP profile used for the SBC 1000/2000 for this testing effort:
Select Settings > SIP > SIP Server Tables
SIP Server Tables contain information about the SIP devices connected to the Ribbon 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.
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:
The Media List shows the selected voice and fax compression codecs and their associated settings.
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).
Select Settings > Message Manipulation > Message Rule Tables
Message Rule Tables are sets of Condition Rules and are applied in SIP Signaling Groups when Message Manipulation is enabled.
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.
|1-1||IP Phone to PSTN Phone, IP Phone disconnect after answer||PASS|
|1-2||IP Phone to PSTN Phone, PSTN Phone busy||PASS|
|1-3||IP Phone to PSTN Phone, PSTN Phone no answer||PASS|
|1-4||IP Phone to PSTN Phone, PSTN Phone disconnect after answer||PASS|
|1-5||PSTN Phone to IP Phone, PSTN Phone disconnect after answer||PASS|
|1-6||PSTN Phone to IP Phone, IP Phone busy||N/A||SfB doesn't send 486 Busy Here|
|1-7||PSTN Phone to IP Phone, IP Phone no answer||PASS|
|1-8||PSTN Phone to IP Phone, IP Phone disconnect after answer||PASS|
|1-9||PSTN Phone to IP Phone, network disconnect||PASS|
|1-11||IP Phone to International Mobile||PASS|
|1-12||IP Phone to International PSTN Phone, remote ringback||PASS|
|1-13||IP Phone to PSTN Phone, Long Duration Call||PASS|
|1-14||IP Phone to PSTN Phone, Mute both ends of call||PASS|
|2-1||PSTN 1 to IP Phone A1, A1 blind transfers to PSTN 2||PASS|
|2-2||PSTN 1 to IP Phone A1, A1 consultative transfers to PSTN 2||PASS|
|2-3||PSTN 1 to IP Phone A1, A1 forwards to PSTN 2, Unconditional||PASS|
|2-4||PSTN 1 to IP Phone A1, A1 forwards to busy PSTN 2, Unconditional||PASS|
|2-5||PSTN 1 to IP Phone A1, A1 forwards to IP Phone A2, Unconditional||PASS|
|2-6||PSTN 1 to IP Phone A1, A1 forwards to Mobile, Unconditional||PASS|
|2-7||PSTN 1 to IP Phone A1, A1 forwards to PSTN 2, No Answer||PASS|
|2-8||PSTN 1 to IP Phone A1, A1 forwards to IP Phone A2, No Answer||PASS|
|2-9||PSTN 1 to IP Phone A1, A1 forwards to Mobile, No Answer||PASS|
|2-10||IP Phone A1 to PSTN 1, A1 conference to PSTN 2, after answer||PASS|
|2-11||IP Phone A1 to PSTN 1, A1 conference to IP Phone A2, after answer||PASS|
|2-12||IP Phone A1 to PSTN 1, A1 conference to IP Phone A2, mixed codecs||PASS|
|4-1||IP Phone to PSTN Phone, G.729 codec||PASS|
|4-2||IP Phone to PSTN Phone, G.711 alaw codec||PASS|
|4-3||PSTN Phone to IP Phone, G.729 codec||PASS|
|4-4||PSTN Phone to IP Phone, G.711 alaw codec||PASS|
|4-5||IP Phone to PSTN Phone, G.726 32K codec||PASS|
|4-6||PSTN Phone to IP Phone, G.726 32K codec||PASS|
|4-7||IP Phone to PSTN Phone, G.711 Ulaw codec||PASS|
|4-8||PSTN Phone to IP Phone, G.711 Ulaw codec||PASS|
|4-9||IP Phone to PSTN Phone, iLBC codec||NOT SUPPORTED||iLBC not supported by both SBC and SfB|
|4-10||PSTN Phone to IP Phone, iLBC codec||NOT SUPPORTED||iLBC not supported by both SBC and SfB|
|4-11||IP Phone to IP Phone, G.722 codec||NOT SUPPORTED||SfB uses SILK for peer to peer if no bandwidth limitations are applied or detected, otherwise will use RTA for low bandwidth|
|5-1||IP Phone to PSTN Phone, DTMF using RFC2833||PASS|
|5-2||PSTN Phone to IP Phone, DTMF using RFC2833||PASS|
|5-3||IP Phone to PSTN Phone, DTMF using H.245 Signal||NOT SUPPORTED||SfB doesn’t support it|
|5-4||PSTN Phone to IP Phone, DTMF using H.245 Signal||NOT SUPPORTED||SfB doesn’t support it|
|5-5||IP Phone to PSTN Phone, DTMF using H.245 Alphanumeric||NOT SUPPORTED||SfB doesn’t support it|
|5-6||PSTN Phone to IP Phone, DTMF using H.245 Alphanumeric||NOT SUPPORTED||SfB doesn’t support it|
|5-7||IP Phone to PSTN Phone, DTMF Before Answer||PASS|
|6-1||Caller ID Presentation (CLIP) with No Screening||PASS|
|6-2||Caller ID Presentation (CLIP) Screening with Correct CLI||PASS|
|6-3||Caller ID Presentation (CLIP) Screening with Incorrect CLI||PASS|
|6-4||IP Phone to PSTN Phone, Caller ID Restriction (CLIR)||PASS|
|6-5||PSTN Phone to IP Phone, Caller ID Restriction (CLIR)||PASS|
|7-1||IP Phone to PSTN Phone, TLS + RTP||PASS|
|7-2||PSTN Phone to IP Phone, TLS + RTP||PASS|
|7-3||IP Phone to PSTN Phone, TLS + SRTP||PASS|
|7-4||PSTN Phone to IP Phone, TLS + SRTP||PASS|
These Application Notes describe the configuration steps required for the Ribbon 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.