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Table of Contents

 

 

Document Overview

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.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flows between Ribbon SBC Edge and Skype for Business 2015.

 

Audience

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.


Requirements

The following equipment and software were used for the sample configuration provided:

 

Equipment

Software Version

Ribbon

SBC 2000

V8.0.0build502

Tenor AFM200P108-09-26

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 Ribbon SBC Edge.

Support

For any questions regarding this document or the content herein, please contact your maintenance and support provider.

 

Third-Party Product Features

The testing was executed with the Colt test plan. The following features were tested:

 

Verify License

SIP Calls


Skype for Business 2015 Configuration

The following new configurations are included in this section:

  1. PSTN Gateway
  2. Voice Policy
  3. PSTN Usage
  4. Route
  5. Trunk Configuration

1. PSTN Gateway

Topology Builder > Shared Components > PSTN Gateways

 

 

 

 

2. Voice Policy

Control Panel > Voice Routing > Voice Policy


 

3. PSTN Usage

Control Panel > Voice Routing > PSTN Usage

 

4. Route

Control Panel > Voice Routing > Route

 

 

5. Trunk Configuration

Control Panel > Voice Routing > Trunk Configuration

 

 

Ribbon SBC 1000/2000 Configuration

The following steps provide an example of how to configure Ribbon SBC 1000/2000:

  1. SIP Profile
  2. SIP Server
  3. Media Profile
  4. Media List
  5. Transformation Table
  6. Call Routing Table
  7. Message Rule Tables
  8. Signaling Groups

1. SIP Profile

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:

 


2. SIP Server

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.

 

 

 


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:

 

 

 

 

 

4. Media List

The Media List shows the selected voice and fax compression codecs and their associated settings.


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).

 

 


7. Message Rule Tables

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.

 


8. 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.

 

 

 

Test Results

 

S.NoProcedureObservationResultComment
Basic Calls
1-1IP Phone to PSTN Phone, IP Phone disconnect after answer PASS 
1-2IP Phone to PSTN Phone, PSTN Phone busy PASS 
1-3IP Phone to PSTN Phone, PSTN Phone no answer PASS 
1-4IP Phone to PSTN Phone, PSTN Phone disconnect after answer PASS 
1-5PSTN Phone to IP Phone, PSTN Phone disconnect after answer PASS 
1-6PSTN Phone to IP Phone, IP Phone busy N/ASfB doesn't send 486 Busy Here
1-7PSTN Phone to IP Phone, IP Phone no answer PASS 
1-8PSTN Phone to IP Phone, IP Phone disconnect after answer PASS 
1-9PSTN Phone to IP Phone, network disconnect PASS 
1-11IP Phone to International Mobile PASS 
1-12IP Phone to International PSTN Phone, remote ringback PASS 
1-13IP Phone to PSTN Phone, Long Duration Call PASS 
1-14IP Phone to PSTN Phone, Mute both ends of call PASS 
Enhanced Calls
2-1PSTN 1 to IP Phone A1, A1 blind transfers to PSTN 2 PASS 
2-2PSTN 1 to IP Phone A1, A1 consultative transfers to PSTN 2 PASS 
2-3PSTN 1 to IP Phone A1, A1 forwards to PSTN 2, Unconditional PASS 
2-4PSTN 1 to IP Phone A1, A1 forwards to busy PSTN 2, Unconditional PASS 
2-5PSTN 1 to IP Phone A1, A1 forwards to IP Phone A2, Unconditional PASS 
2-6PSTN 1 to IP Phone A1, A1 forwards to Mobile, Unconditional PASS 
2-7PSTN 1 to IP Phone A1, A1 forwards to PSTN 2, No Answer PASS 
2-8PSTN 1 to IP Phone A1, A1 forwards to IP Phone A2, No Answer PASS 
2-9PSTN 1 to IP Phone A1, A1 forwards to Mobile, No Answer PASS 
2-10IP Phone A1 to PSTN 1, A1 conference to PSTN 2, after answer PASS 
2-11IP Phone A1 to PSTN 1, A1 conference to IP Phone A2, after answer PASS 
2-12IP Phone A1 to PSTN 1, A1 conference to IP Phone A2, mixed codecs PASS 
Codec Support
4-1IP Phone to PSTN Phone, G.729 codec PASS 
4-2IP Phone to PSTN Phone, G.711 alaw codec PASS 
4-3PSTN Phone to IP Phone, G.729 codec PASS 
4-4PSTN Phone to IP Phone, G.711 alaw codec PASS 
4-5IP Phone to PSTN Phone, G.726 32K codec PASS 
4-6PSTN Phone to IP Phone, G.726 32K codec PASS 
4-7IP Phone to PSTN Phone, G.711 Ulaw codec PASS 
4-8PSTN Phone to IP Phone, G.711 Ulaw codec PASS 
4-9IP Phone to PSTN Phone, iLBC codec NOT SUPPORTEDiLBC not supported by both SBC and SfB
4-10PSTN Phone to IP Phone, iLBC codec NOT SUPPORTEDiLBC not supported by both SBC and SfB
4-11IP Phone to IP Phone, G.722 codec  NOT SUPPORTEDSfB uses SILK for peer to peer if no bandwidth limitations are applied or detected, otherwise will use RTA for low bandwidth
DTMF Support
5-1IP Phone to PSTN Phone, DTMF using RFC2833 PASS 
5-2PSTN Phone to IP Phone, DTMF using RFC2833 PASS 
5-3IP Phone to PSTN Phone, DTMF using H.245 Signal NOT SUPPORTEDSfB doesn’t support it
5-4PSTN Phone to IP Phone, DTMF using H.245 Signal NOT SUPPORTEDSfB doesn’t support it
5-5IP Phone to PSTN Phone, DTMF using H.245 Alphanumeric NOT SUPPORTEDSfB doesn’t support it
5-6PSTN Phone to IP Phone, DTMF using H.245 Alphanumeric NOT SUPPORTEDSfB doesn’t support it
5-7IP Phone to PSTN Phone, DTMF Before Answer PASS 
CLI Services
6-1Caller ID Presentation (CLIP) with No Screening PASS 
6-2Caller ID Presentation (CLIP) Screening with Correct CLI PASS 
6-3Caller ID Presentation (CLIP) Screening with Incorrect CLI PASS 
6-4IP Phone to PSTN Phone, Caller ID Restriction (CLIR) PASS 
6-5PSTN Phone to IP Phone, Caller ID Restriction (CLIR) PASS 
Encryption
7-1IP Phone to PSTN Phone, TLS + RTP PASS 
7-2PSTN Phone to IP Phone, TLS + RTP PASS 
7-3IP Phone to PSTN Phone, TLS + SRTP PASS 
7-4PSTN Phone to IP Phone, TLS + SRTP PASS 

 

Conclusion

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.

 

Appendix A - TLS/SRTP Configuration