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

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1 SBC Edge 2000 V5.0.1 IOT Skype for Business 2015 Intermedia SIP Trunk Application Notes Document Overview Introduction Audience Requirements Reference Configuration Support Third-party Product Features Requirements Verify License SFB2015 Configuration Overview PSTN Gateway Voice Policy PSTN Usage Route Trunk Configuration SBC Edge 2000 Configuration Overview SIP Profile SIP Server Media Profile Media List Contact Registrant Tables Remote Authorization Tables Signaling Groups Transformation Call Routing Table Test Results Conclusion Document Overview This document provides a configuration guide for Sonus SBC Edge Series (Session Border Controller) when connecting 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 SFB2015, please visit For additional information on Sonus SBC Edge Series, please visit Introduction The interoperability compliance testing focuses on verifying various inbound and outbound calls flows between Sonus SBC Edge series and SFB2015 Audience This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus Copyright 2016 Sonus Networks. All rights reserved. Page 1

2 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 for 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 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: Equipment Version Sonus Networks SBC Edge 2000 V5.0.1build399 Third Party Equipments SFB Reference Configuration The following reference configuration shows connectivity between third-party and Sonus SBC Edge. Copyright 2016 Sonus Networks. All rights reserved. Page 2

3 Support Technical support on Sonus SBC Edge series can be obtained through the following: Phone: (978) or (888) (Toll-free) Web: Third-party Product Features The testing was executed with the Intermedia test plan, and the following features were tested: Basic originated and terminated calls Basic inbound/outbound call Hold and Resume Call Forwarding Unconditional FAX DTMF Requirements Sonus Equipment Type Version SBC Edge SBC b399 3rd Party Equipment Type Version Microsoft Skype for Business 2015 Mediation Server Polycom CX600 SIP Phone Verify License No special licensing required. SFB2015 Configuration Overview New PSTN gateway New Voice Policy New PSTN Usage New Route New Trunk Configuration Copyright 2016 Sonus Networks. All rights reserved. Page 3

PSTN Gateway Topology Builder > Shared Components > PSTN

Voice Policy Control Panel > Voice Routing > Voice Policy

Route Control Panel > Voice Routing > Route

Trunk Configuration Control Panel > Voice Routing > Trunk

SBC Edge 2000 Configuration Overview 1. 2. 3. 4. 5. 6. 7. 8. 9.

13 SBC Edge 2000 Configuration Overview SIP Profile SIP Server Media Profile Media Lists Contact Registrant Tables Remote Authorization Tables Signaling Group Transformation Call Routing Table SIP Profile Select Settings > SIP > SIP Profiles SIP Profiles control how the Sonus SBC Edge 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 Edge for this testing effort is shown below. Copyright 2016 Sonus Networks. All rights reserved. Page 13

14 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 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. Copyright 2016 Sonus Networks. All rights reserved. Page 14

bandwidth requirements at the expense of voice quality.

15 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 Edge in this testing effort and is for reference only. Copyright 2016 Sonus Networks. All rights reserved. Page 15

* 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.

16 * 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 Server Configuration can specify a Contact Registrant Table, and the username portion of the table will be used for outbound calls. Copyright 2016 Sonus Networks. All rights reserved. Page 16

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

Signaling Groups Select Settings > Signaling Groups Signaling Groups allow telephony channels to be grouped together for the purposes of routing and shared configuration.

17 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. 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. Copyright 2016 Sonus Networks. All rights reserved. Page 17

Transformation Select Settings > Transformation Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call.

19 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. Copyright 2016 Sonus Networks. All rights reserved. Page 19

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

20 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 Copyright 2016 Sonus Networks. All rights reserved. Page 20

21 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). Copyright 2016 Sonus Networks. All rights reserved. Page 21

22 Test Results S.No Procedure Observation Result Comment 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 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 2.2 Normal Call Customer initiate call, AccessLine hang-up 180/183 + Answer BYE from AccessLine Copyright 2016 Sonus Networks. All rights reserved. Page 22

23 2.3 No answer call 180/183 + CANCEL Customer initiate call, Customer hang-up before answer 2.4 Busy call Customer calls a busy number 180/ The calling party should play busy tone to the caller. 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 min call Customer initiate call 180/183 + Answer AccessLine SBC doesn t use session timer, it does detect RTP timeout 2.7 Fax/Modem call Customer initiate call Test with both G729 and G /183 + Answer with voice coders If the initial codec is G729, Accessline sends a reinvite with G Anonymous From 180/183 + Answer Customer initiate call with From=Anonymous 2.9 G /183 + Answer with G.729 Customer initiate call with SDP=G.729, G.711 DTMF RFC2833 in G way DTMF 2.10 G /183 + Answer with G.711 Customer initiate call with SDP=G.711, G.729 DTMF RFC2833 in G way DTMF 2.11 AccessLine changes codecs after answer Customer initiate call with SDP=G.729 and G AccessLine sends 180/183/200 SDP=G AccessLine sends a reinvite with SDP=G Customer changes codecs after answer Customer initiate call with SDP=G.729 and G AccessLine sends 180/183/200 SDP=G Customer sends a reinvite with SDP=G.711 N/A SFB does not supported it 2.13 RTCP Check RTCP from both AccessLine and customer Copyright 2016 Sonus Networks. All rights reserved. Page 23

24 3.1 Normal Call AccessLine initiate call, AccessLine hang-up 180/183 + Answer BYE from AccessLine 3.2 Normal Call AccessLine initiate call, Customer hang-up 180/183 + Answer BYE from Customer 3.3 No answer call 180/183 + CANCEL AccessLine initiate call, AccessLine hang-up before answer 3.4 Busy call AccessLine calls a busy number 3.5 Not Found 180/ N/A SFB does not supported it 404 or any 4xx/5xx/6xx AccessLine calls a non routable number min call AccessLine initiate call 180/183 + Answer AccessLine SBC doesn t use session timer, it does detect RTP timeout 3.7 Fax/modem call AccessLine initiate call Test with both G729 and G /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 G G.729 AccessLine initiate call with SDP=G.729, G.711 DTMF RFC2833 in G /183 + Answer 2 way DTMF 3.9 G.711 AccessLine initiate call with SDP=G.711, G.729 DTMF RFC2833 in G /183 + Answer 2 way DTMF 3.10 Anonymous From 180/183 + Answer AccessLine initiate call with From=Anonymous Copyright 2016 Sonus Networks. All rights reserved. Page 24

25 3.11 AccessLine changes codec after answer AccessLine initiate call with SDP=G Customer sends 180/183/200 SDP=G AccessLine sends reinvite with SDP=G Customer changes codec after answer AccessLine initiate call with SDP=G Customer sends 180/183/200 SDP=G Customer sends reinvite with SDP=G.729 N/A SFB does not supported it 3.13 RTCP Check RTCP from both AccessLine and customer 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 supported 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 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 4.4 Local Conference The PBX mixes the audio from a few calls 4.5 Call Reject The PBX rejects a call with 4xx status 4.6 Hold / Unhold The PBX sends a reinvite with hold indication Conclusion These Application Notes describe the configuration steps required for Sonus SBC Edge series to successfully interoperate with Skype for Business All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results. Copyright 2016 Sonus Networks. All rights reserved. Page 25

Sonus SBC 1000/2000 V6.0.0 Registration IOT Skype for Business 2015 Analog Devices and Failover Application Note

Sonus SBC 1000/2000 V6.0.0 Registration IOT Skype for Business 2015 Analog Devices and Failover Application Note Table of Contents Copyright 2017 Sonus Networks. All rights reserved. Page 1 Document Overview