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Document Overview

This document provides a configuration guide for Sonus SBC 2000 (Session Border Controller) when connecting to Skype for Business 2015 (SFB2015) and Intermedia SIP Trunk.

This configuration guide supports features provided in Microsoft Technet web page.

Introduction

The interoperability compliance testing focuses on verifying various inbound and outbound calls flows between Sonus SBC 2000 and SFB2015.

Audience

This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus SBC 2000 aspects of the Intermedia SIP trunk 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:

 EquipmentVersion
Sonus NetworksSBC 2000 V5.0.1build399
Third-Party EquipmentMicrosoft Skype for Business 2015 (SFB2015) Mediation Server 6.0.9319.0

Polycom CX600 SIP Phone

4.0.7577.44455

Reference Configuration

The following reference configuration shows connectivity between third-party and Sonus SBC 2000.

Figure : Connectivity Between Third-Party and Sonus SBC 2000

 

Support

Technical support for Sonus SBC 2000 series is available via phone or logging a trouble ticket:

Third-Party Product Features

The following features were tested using the Intermedia test plan:

  • Basic originated and terminated calls
  • Basic inbound/outbound call
  • Hold and Resume
  • Call Forwarding Unconditional
  • FAX
  • DTMF 

Verify License

No special licensing required.

 

SFB2015 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

Figure : Define a new IP/PSTN Gateway

Figure : Define FQDN

Figure : Define IP Address

Figure : Define Root Trunk

2. Voice Policy

Control Panel > Voice Routing > Voice Policy

Figure : Edit Voice Policy

3. PSTN Usage

Control Panel > Voice Routing > PSTN Usage

Figure : View PSTN Usage

4. Route

Control Panel > Voice Routing > Route

Figure : Edit Voice Route

 

5. Trunk Configuration

Control Panel > Voice Routing > Trunk Configuration

Figure : Edit Trunk Configuration

Sonus SBC 2000 Configuration

The following steps provide an example of how to configure Sonus SBC 2000.

 

  1. SIP Profile
  2. SIP Server
  3. Media Profile
  4. Media Lists
  5. Contact Registrant Tables
  6. Remote Authorization Tables
  7. Signaling Groups
  8. Transformation
  9. Call Routing Table


1. SIP Profile

Select Settings > SIP > SIP Profiles

SIP Profiles control how the Sonus SBC 2000 communicates with SIP devices. These control important characteristics such as: session timers, SIP Header customization, SIP timers, MIME payloads, and option tags. The default SIP profile used for the SBC 2000 for this testing effort is shown below.

Figure : Default SIP Profile

 

2. SIP Server 

Select Settings > SIP > SIP Server Tables 

SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 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.

 

Figure : SIP Server

 

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. Listed below are the media profiles of the voice codecs used for the SBC 2000 in this testing effort and is for reference only.

Figure : Voice Codec Media Profiles

SFB uses default Media List

4. Media Lists

Select Settings > Media > Media List

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

Figure : Media Lists


5. Contact Registrant Tables 

Select Settings > SIP > Contact Registrant Tables

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 will be used for outbound calls.

Figure : Contact Registrant Table Example

6. Remote Authorization Tables

 Select Settings > SIP > 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 defined in this page appear as options in the Remote Authorization and Contacts Panel for SIP Servers.

Figure : Remote Authorization Table Example


7. 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.

Figure : Signaling Groups Example

8. Transformation

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 will be configurable as a reusable pool that Action Sets can reference.

Figure : Transformation Tables Example

 

9. 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).

Figure : Call Routing Table


Test Results

S.NoProcedureObservationResultComment

1.1

Customer send REGISTER and accept 401 response for authentication

AccessLine may set the Registration expiration time to 30 seconds in order to refresh NAT router binding

Passed

 

1.2

If customer has other means of NAT router refresh messages, such as OPTIONS, STUN etc.

 

N/A

 

2.1

Normal Call

Customer initiate call, Customer hang-up

180/183 + Answer

BYE from customer

Passed

 

2.2

Normal Call

Customer initiate call, AccessLine hang-up

180/183 + Answer

BYE from AccessLine

Passed

 

2.3

No answer call

Customer initiate call,

Customer hang-up before answer

180/183 + CANCEL

Passed

 

2.4

Busy call

Customer calls a busy number

180/183 + 486
The calling party should play busy tone to the caller.

Passed

 

2.5

Not Found

Customer calls a non routable number

404/480 or any other 4xx/5xx/6xx

The calling party should play fast busy to the caller.

Passed

 

2.6

10 min call

Customer initiate call

180/183 + Answer

AccessLine SBC doesn’t use session timer, it does detect RTP timeout

Passed

 

2.7

Fax/Modem call

Customer initiate call

Test with both G729 and G711

180/183 + Answer with voice coders

If the initial codec is G729, Accessline sends a reINVITE with G711

Passed

 

2.8

Anonymous From

Customer initiate call with

From=Anonymous

180/183 + Answer

Passed

 

2.9

G.729

Customer initiate call with SDP=G.729 , G.711

DTMF RFC2833 in G.729

180/183 + Answer with G.729

 

2 way DTMF

Passed

 

2.10

G.711

Customer initiate call with SDP=G.711 , G.729

DTMF RFC2833 in G.711

180/183 + Answer with G.711

 

2 way DTMF

Passed

 

2.11

AccessLine changes codecs after answer

Customer initiate call with SDP=G.729 and G.711

1. AccessLine sends 180/183/200 SDP=G.729

2. AccessLine sends a reINVITE with SDP=G.711

 

Passed

 

2.12

Customer changes codecs after answer

Customer initiate call with SDP=G.729 and G.711

1. AccessLine sends 180/183/200 SDP=G.729

2. Customer sends a reINVITE with SDP=G.711

N/A

SFB does not support it

2.13

RTCP

Check RTCP from both AccessLine and customer

Passed

 

3.1

Normal Call

AccessLine initiate call, AccessLine hang-up

180/183 + Answer

BYE from AccessLine

Passed

 

3.2

Normal Call

AccessLine initiate call, Customer hang-up

180/183 + Answer

BYE from Customer

Passed

 

3.3

No answer call

AccessLine initiate call,

AccessLine hang-up before answer

180/183 + CANCEL

Passed

 

3.4

Busy call

AccessLine calls a busy number

180/183 + 486

N/A

SFB does not support it

3.5

Not Found

AccessLine calls a non routable number

404 or any 4xx/5xx/6xx

Passed

 

3.6

10 min call

AccessLine initiate call

180/183 + Answer

AccessLine SBC doesn’t use session timer, it does detect RTP timeout

Passed

 

3.7

Fax/modem call

AccessLine initiate call

Test with both G729 and G711

180/183 + Answer with voice coders

If the initial codec is G729, the customer should detect the fax/modem answering tone and send a reINVITE with G711.

Passed

 

3.8

G.729

AccessLine initiate call with SDP=G.729 , G.711

DTMF RFC2833 in G.729

180/183 + Answer

 

2 way DTMF

Passed

 

3.9

G.711

AccessLine initiate call with SDP=G.711 , G.729

DTMF RFC2833 in G.711

180/183 + Answer

 

2 way DTMF

Passed

 

3.10

Anonymous From

AccessLine initiate call with From=Anonymous

180/183 + Answer

Passed

 

3.11

AccessLine changes codec after answer

AccessLine initiate call with SDP=G.729

1. Customer sends 180/183/200 SDP=G.729

2. AccessLine sends reinvite with SDP=G.711

Passed

 

3.12

Customer changes codec after answer

AccessLine initiate call with SDP=G.711

1. Customer sends 180/183/200 SDP=G.711

2. Customer sends reinvite with SDP=G.729

N/A

SFB does not support it

3.13

RTCP

Check RTCP from both AccessLine and customer

Passed

 

4.1

Call forward on busy

There are two possible implementations:

  • PBX makes an outbound call and links the audio between the calls
  • PBX sends 3xx response with destination in Contact header

N/A

SFB does not support it

4.2

Call forward on no answer

There are two possible implementations:

  • PBX makes an outbound call and links the audio between the calls
  • PBX sends 3xx response with destination in Contact header

Passed

 

4.3

Call transfer

There are two possible implementations:

  • PBX makes an outbound call and links the audio between the calls
  • PBX makes an outbound call and sends SIP REFER method with Refer-To  header containing a Replaces field

Passed

 

4.4

Local Conference

The PBX mixes the audio from a few calls

Passed

 

4.5

Call Reject

The PBX rejects a call with 4xx status

Passed

 

4.6

Hold / Unhold

The PBX sends a reINVITE with hold indication

Passed

 

 

Conclusion

This Application Notes document describes the configuration steps required for Sonus SBC 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.