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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.

For additional information on Skype for Business 2015, please visit http://microsoft.com
For additional information on Ribbon SBC Edge Series (Session Border Controller), please visit https://ribboncommunications.com/

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.

Info

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:

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0	Table
1	Requirements

	Equipment	Software Version
Ribbon	SBC 2000	V8.0.0build502
Ribbon	Tenor AFM200	P108-09-26
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.

Caption

0	Figure
1	Connectivity Between Third-Party and Ribbon SBC 2000
3	Connectivity Between Third-Party and Ribbon SBC 2000

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:

Basic Calls
Enhanced Calls
Codec Support
DTMF Support
CLI Services
Encryption

Verify License

SIP Calls

Skype for Business 2015 Configuration

The following new configurations are included in this section:

Anchor
PSTN Gateway
PSTN Gateway
1. PSTN Gateway

Topology Builder > Shared Components > PSTN Gateways

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0	Figure
1	Define a new IP/PSTN Gateway
3	Define a new IP/PSTN Gateway

Noprint

Caption

0	Figure
1	Define FQDN
3	Define FQDN

Noprint

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0	Figure
1	Define IP Address
3	Define IP Address

Noprint

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0	Figure
1	Define Root Trunk
3	Define Root Trunk

Noprint

Anchor
Voice Policy
Voice Policy
2. Voice Policy

Control Panel > Voice Routing > Voice Policy

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0	Figure
1	Edit Voice Policy
3	Edit Voice Policy

Noprint

Anchor
PSTN Usage
PSTN Usage
3. PSTN Usage

Control Panel > Voice Routing > PSTN Usage

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0	Figure
1	View PSTN Usage
3	View PSTN Usage

Noprint

Anchor
Route
Route
4. Route

Control Panel > Voice Routing > Route

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0	Figure
1	Edit Voice Route
3	Edit Voice Route

Noprint

Anchor
Trunk Configuration
Trunk Configuration
5. Trunk Configuration

Control Panel > Voice Routing > Trunk Configuration

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0	Figure
1	Edit Trunk Configuration
3	Edit Trunk Configuration

Ribbon SBC 1000/2000 Configuration

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

Anchor
Step-1
Step-1
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:

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0	Figure
1	SIP Profiles
3	SIP Profiles

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Step-2
Step-2
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.

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0	Figure
1	Skype
3	Skype

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0	Figure
1	Fax
3	Fax

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0	Figure
1	Colt
3	Colt

Anchor
Step-3
Step-3
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:

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0	Figure
1	Voice Codec G711 A-Law
3	Voice Codec G711 A-Law

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0	Figure
1	Voice Codec G711 U-Law
3	Voice Codec G711 U-Law

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0	Figure
1	Voice Codec G729
3	Voice Codec G729

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0	Figure
1	T.38
3	T.38

Anchor
Step-4
Step-4
4. Media List

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

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0	Figure
1	Media List
3	Media Lists

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0	Figure
1	From Skype
3	From Skype

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Step-6
Step-6
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).

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0	Figure
1	From Colt
3	From Colt

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0	Figure
1	From Skype to Colt
3	From Skype to Colt

Anchor
Step-7
Step-7
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.

Caption

0	Figure
1	Colt Outbound
3	Colt Outbound

Anchor
Step-8
Step-8
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.

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0	Figure
1	Internal Side
3	Internal Side

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0	Figure
1	External Side
3	External Side

Anchor
Test Results
Test Results
Test Results

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0	Table
1	Test Results
3	Test Results

S.No	Procedure	Result	Comment
Basic Calls
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
Enhanced Calls
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
Codec Support
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
DTMF Support
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
CLI Services
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
Encryption
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

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

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1	TLS Profile
3	TLS Profile

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1	SRTP Profile
3	SRTP Profile

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0	Figure
1	Colt SIP Server
3	Colt SIP Server

Caption

0	Figure
1	Colt Media List
3	Colt Media List

Caption

0	Figure
1	Colt Signaling Group
3	Colt Signaling Group

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