SBC Edge 2000 V5.0.1 IOT Skype for Business 2015 ... 0... · Copyright © 2016 Sonus Networks. All rights reserved. Page 23 2.3 No answer call Customer initiate call, Customer hang-up

Copyright © 2016 Sonus Networks. All rights reserved. Page 1

SBC Edge 2000 V5.0.1 IOT Skype for Business 2015Intermedia SIP Trunk Application Notes

Document OverviewIntroduction

AudienceRequirementsReference ConfigurationSupportThird-party Product FeaturesRequirementsVerify License

SFB2015 ConfigurationOverviewPSTN GatewayVoice PolicyPSTN UsageRouteTrunk Configuration

SBC Edge 2000 ConfigurationOverviewSIP ProfileSIP Server Media ProfileMedia ListContact Registrant Tables Remote Authorization TablesSignaling GroupsTransformationCall Routing Table

Test ResultsConclusion

Document Overview

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

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

For additional information on , please visit .comSFB2015 http://microsoftFor additional information on Sonus SBC Edge Series, please visit http://sonus.net.

Introduction

The interoperability compliance testing focuses on verifying various inbound and outbound calls flows betweenSonus SBC and Edge series SFB2015

Audience

This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus

http://microsoft

http://sonus.net.


SBC Edge series aspects of the Intermedia SIP trunk group together with the SFB2015 . There will be steps that require navigating a third-party and Sonus SBC Web browser user interface. Understanding the basic concepts forIP/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 informationregarding the product in this guide are subject to change without notice. All statements, information, andrecommendations in this guide are believed to be accurate but are presented without warranty of any kind, expressor implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications andinformation in this guide.

Requirements

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

Equipment Version

Sonus Networks

SBC Edge 2000 V5.0.1build399

Third Party Equipments

SFB2015 6.0.9319.0

Reference Configuration

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


1. 2. 3. 4. 5.

Support

Technical support on Sonus SBC Edge series can be obtained through the following:

Phone: (978) 614-8589 or (888) 391-3434 (Toll-free)Web: http://sonusnetworks.force.com/PortalLoginPage

Third-party Product Features

The testing was executed with the Intermedia test plan, and the following features were tested:

Basic originated and terminated callsBasic inbound/outbound callHold and ResumeCall Forwarding UnconditionalFAXDTMF

Requirements

Sonus Equipment Type Version

SBC Edge SBC 2000 5.0.1b399

3rd Party Equipment Type Version

Microsoft Skype for Business 2015 Mediation Server 6.0.9319.0

Polycom CX600 SIP Phone 4.0.7577.44455

Verify License

No special licensing required.

SFB2015 Configuration

OverviewNew PSTN gatewayNew Voice PolicyNew PSTN UsageNew RouteNew Trunk Configuration

http://sonusnetworks.force.com/PortalLoginPage


5.

PSTN Gateway

Topology Builder > Shared Components > PSTN Gateways




Voice Policy

Control Panel > Voice Routing > Voice Policy


PSTN Usage

Control Panel > Voice Routing > PSTN Usage


Route

Control Panel > Voice Routing > Route



Trunk Configuration

Control Panel > Voice Routing > Trunk Configuration



1. 2. 3. 4. 5. 6. 7. 8. 9.

SBC Edge 2000 Configuration

OverviewSIP ProfileSIP ServerMedia ProfileMedia ListsContact Registrant TablesRemote Authorization TablesSignaling GroupTransformationCall Routing Table

SIP Profile

Select Settings > SIP > SIP Profiles

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


SIP Server

Select Settings > SIP > SIP Server Tables

SIP Server Tables contain information about the SIP devices connected to the Sonus SBC Edge. The entries in thetables 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.


Media Profile

Select Settings > Media > Media Profiles

Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusioninto a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidthrequirements at the expense of voice quality. Listed below are the media profiles of the voice codecs used for theSBC Edge in this testing effort and is for reference only.


* SFB uses default Media List

Media List

Select Settings > Media > Media List

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

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 ServerConfiguration can specify a Contact Registrant Table, and the username portion of the table will be used foroutbound calls.


Remote Authorization Tables

Select Settings > SIP > Remote Authorization Tables

Remote Authorization Tables and their entries contain information used to respond to request message challengesby an upstream server. The Remote Authorization tables defined in this page appear as options in the Remote

for SIP Servers.Authorization and Contacts Panel

Signaling Groups

Select Settings > Signaling Groups

Signaling Groups allow telephony channels to be grouped together for the purposes of routing and sharedconfiguration. These groups are the entity to which calls are routed, as well as the location from which Call Routesare 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.

http://wiki.sonusnet.com/display/UXDOC50/Creating+and+Modifying+Entries+in+SIP+Server+Tables#CreatingandModifyingEntriesinSIPServerTables-AddinganEntrytoaSIPServerTable

http://wiki.sonusnet.com/display/UXDOC50/Creating+and+Modifying+Entries+in+SIP+Server+Tables#CreatingandModifyingEntriesinSIPServerTables-AddinganEntrytoaSIPServerTable



Transformation

Select Settings > Transformation

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. Forexample, 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 fromthere. In addition, Transformation tables will be configurable as a reusable pool that Action Sets can reference.


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, andbetween 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 centralconnection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, MediaLists and the three types of Signaling Groups (ISDN, SIP and CAS).


Test Results

S.No Procedure Observation Result Comment

1.1 Customer send REGISTER and accept401 response for authentication

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

Passed

1.2 If customer has other means of NATrouter refresh messages, such asOPTIONS, STUN etc.

N/A

2.1 Normal Call

Customer initiate call, Customerhang-up

180/183 + Answer

BYE from customer

Passed

2.2 Normal Call

Customer initiate call, AccessLinehang-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 + 486The 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 doesdetect 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 areINVITE 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


180/183 + Answer with G.711

2 way DTMF

Passed

2.11 AccessLine changes codecs afteranswer

Customer initiate call with SDP=G.729and 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 afteranswer

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

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

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

N/A SFB doesnotsupportedit

2.13 RTCP Check RTCP from both AccessLine and customer Passed


3.1 Normal Call

AccessLine initiate call, AccessLinehang-up

180/183 + Answer

BYE from AccessLine

Passed

3.2 Normal Call

AccessLine initiate call, Customerhang-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 doesnotsupportedit

3.5 Not Found

AccessLine calls a non routablenumber

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 doesdetect 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 shoulddetect the fax/modem answering tone and send areinvite with G711.

Passed

3.8 G.729

AccessLine initiate call withSDP=G.729 , G.711


180/183 + Answer

2 way DTMF

Passed

3.9 G.711

AccessLine initiate call withSDP=G.711 , G.729


180/183 + Answer

2 way DTMF

Passed

3.10 Anonymous From

AccessLine initiate call withFrom=Anonymous

180/183 + Answer Passed


3.11 AccessLine changes codec afteranswer

AccessLine initiate call withSDP=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 withSDP=G.711

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

2. Customer sends reinvite with SDP=G.729


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 theaudio between the calls

PBX sends 3xx response with destination inContact header


4.2 Call forward on no answer There are two possible implementations:

PBX makes an outbound call and links theaudio between the calls

PBX sends 3xx response with destination inContact header

Passed

4.3 Call transfer There are two possible implementations:

PBX makes an outbound call and links theaudio between the callsPBX makes an outbound call and sends SIPREFER method with Refer-To headercontaining 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

These Application Notes describe the configuration steps required for Sonus SBC Edge series to successfullyinteroperate with Skype for Business 2015. All feature and serviceability test cases were completed and passed withthe exceptions/observations noted in Test Results.

Documents

SBC Edge 2000 V5.0.1 IOT Skype for Business 2015 ... 0... · Copyright © 2016 Sonus Networks. All rights reserved. Page 23 2.3 No answer call Customer initiate call, Customer hang-up