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Overview

This Application Note is a configuration guide for the Sonus SBC (Session Border Controller) 1000/2000 when connecting to Skype for Business 2015 (Skype 2015) and Virgin Media SIP Trunk.

The configuration guide supports features outlined in the Microsoft Technet web page. 

Introduction

Interoperability compliance testing focuses on verifying inbound and outbound call flows between Sonus SBC 1000/2000 and Skype for Business 2015.

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. There are steps that require navigating third-party equipment as well as the Sonus SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are necessary to complete the configuration and perform any troubleshooting.

This Application Note is offered as a convenience to Sonus customers. The specifications and information regarding the product in this document are subject to change without notice. All statements, information, and recommendations contained in this document 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 contained herein

Requirements

The following table lists the hardware and software used in the reference configuration.

Table : Test Equipment and Software

VendorEquipmentSoftware Version
Sonus NetworksSBC 2000 V5.0.3build407
Tenor AF P108-09-21
Third-party Vendor
MicrosoftMicrosoft Skype for Business 2015 (Skype 2015) Mediation Server 6.0.9319.0

Polycom

Polycom CX600 SIP Phone

 4.0.7577.44455

Reference Configuration

The following figure is a topology for the reference configuration showing connectivity between third-party equipment and the Sonus SBC 1000/2000.

Figure : Reference Configuration Topology

Support

Technical support on Sonus SBC 1000/2000 can be obtained through the following:

Third-Party Product Features

The following third-party product features are supported:

  • Basic originated and terminated calls
  • Basic inbound/outbound call
  • Hold and Resume
  • Call Forwarding
  • FAX
  • DTMF
  • Conference Call
  • Action on eSBC outage (loss of Ethernet , restart of eSBC)
  • Action on Loss of Virgin Media primary SBC

Verify License

No special licensing is required for this test.

 

Skype 2015 Configuration

The following configuration steps are provided to configure Skype 2015 to interoperate with the Sonus SBC 1000/2000:

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

1. PSTN Gateway

Configure the PSTN Gateway using the following configuration screens:

Figure : Define a new IP/PSTN Gateway

Figure : Define FQDN

Figure : Define IP Address Type

Figure : Define Root Trunk

2. Voice Policy

Select Control Panel > Voice Routing > Voice Policy to access the Voice Policy configuration screen. 

Figure : Voice Policy

3. PSTN Usage

Select Control Panel > Voice Routing > PSTN Usage to access the PSTN Usage configuration screen.

Figure : PSTN Usage

4. Route

Select Control Panel > Voice Routing >Route to access the Route configuration screen.

Figure : Route

5. Trunk Configuration

Select Control Panel > Voice Routing >Trunk Configuration to access the trunk configuration screen.

Figure : Trunk Configuration

Sonus SBC 1000/2000 Configuration

The following configuration steps provide an example of how to configure the Sonus SBC 1000/2000 to interoperate with Skype 2015 and Virgin Media SIP Trunk:

  1. SIP Profile
  2. SIP Server
  3. Media System
  4. Media Profiles
  5. Media List
  6. Remote Authorization Tables
  7. Signaling Groups
  8. Transformation
  9. Call Routing Table 
  10. Message Translation
  11. Cause Code Reroute

 

1. SIP Profile

SIP Profiles control how the Sonus 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. 

Select Settings > SIP > SIP Profiles to access the SIP Profile screen.

The default SIP profile used for the SBC 1000/2000 for this testing effort is provided in the following figures.

Figure : Virgin Media SIP Profile

Figure : Skype 2015 SIP Profile

Figure : Fax SIP Profile

2. SIP Server

SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000. 

Select Settings > SIP > SIP Server Tables to access the SIP Server Tables screen.

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, as shown in the following figures.

Figure : Virgin Media SIP Servers

Figure : Skype 2015 SIP Server

Figure : Fax SIP Server

3. Media System

The Media System Configuration contains system-wide settings for the Media System, configuring the media system means setting the number of RTP/RTCP port pairs and the starting port.

Select Settings > Media > Media System Configuration to access the Media System configuration screen.

Figure : Media System

4. 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. 

Select Settings > Media > Media Profiles.

Shown in the following figures are the media profiles of the voice codecs used for the SBC 1000/2000 in this testing effort and are provide for reference only.

Figure : Virgin Media Media Profile

Figure : Skype 2015 Media Profile

Figure : Fax Media Profile

5. Media List

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

 Select Settings > Media > Media List to access the Media List configuration screen.

Figure : Virgin Media Media List

Figure : Skype 2015 Media List

Figure : Fax Media List

6. 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.

Select Settings > SIP > Remote Authorization Tables to access the Remote Authorization Tables configuration screen.

Figure : Remote Authorization Table

 

7. 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.

Select Settings > Signaling Groups to access the Signaling Groups configuration screens.

Figure : Virgin Media Signaling Group

Figure : Skype 2015 Signaling Group

Figure : Fax Signaling Group

8. 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.

Select Settings > Transformation to access the Transformation configuration screen.

Figure : Virgin Media Tranformation

Figure : Skype Transformation

9. 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).

Select Settings > Call Routing Table to access the Call Routing Table configuration screen.

Figure : Virgin Media Call Routing

Figure : Skype Call Routing

Figure : Fax Call Routing

 

10. Message Translation

Message Translation Tables aid in the interworking of differing protocols (like ISDN to SIP) by allowing control over how protocol messages are translated when calls are routed. They are useful for interworking with non-standard equipment and for specialized call routing.

Select Settings > Telephony Mapping Tables > Message Translation to access the Message Translation configuration screen.

Figure : Message Translation

11. Cause Code Reroute

Terminating any calls return a Q.850 Cause Code when they end. We can use these codes to determine whether or not to reroute the call to another signalling group. A Cause Code Reroute table contains one or more Q.850 Cause Codes which, when matched, trigger a reroute.

Select Settings > Telephony Mapping Tables > Cause Code Reroute to access the Cause Code Reroute configuration screen.

Figure : Virgin Media Cause Code Reroute

Interoperability Test Results

The following table provides test results for interoperability compliance testing between Sonus SBC 1000/2000 and Skype for Business 2015.

Table : Interoperability Compliance Test Results

Test NumberTest ScenarioSetup / Test ResultsStatusComment
IOP1vendors eSBC response to SIP OPTIONS messages from SBCNo calls are required for this test. SIP trace to be captured for approx 60 seconds and checked for correct signalling.

For each eSBC, the SBC will periodically send an OPTIONS request to the vendors eSBC to check if its SIP stack is reachable. If a SIP response 200 OK is received from the IP-PBX, the SIP trunk will be placed (or remain) in an In-Service state

e.g. OPTIONS sip:ping@<ip-pbx_IP_Addr>:5060 SIP/2.0
Pass 
IOP2SBC response to SIP OPTIONS messages from vendor eSBCNo calls are required for this test. SIP trace to be captured for approx 60 seconds (depending on agreement) and checked for correct signalling.

Vendors eSBC setup for Solution IP.Addr Mode
eSBC configured to send OPTIONS messages to the SBC on a periodic basis. The SBC responds with SIP response 200OK -
e.g. "OPTIONS sip:ping@192.168.1.10:5060 SIP/2.0"

Check that the eSBC can simultaneously send SIP OPTIONS messages to both the solution SBC addresses.
Pass 
IOP4Basic test call from IP-PBX to PSTN line through SBC-A (using SBC-A IPV4 ip address).IP-PBX line initiates call, Call is answered, IP-PBX line terminates call.

Vendors eSBC setup for Solution IP.Addr Mode
Call from the IP-PBX. Invite seen from eSBC to SBC-A, proxy authentication challenge returned to eSBC, re-invite with correct credentials from eSBC and call progresses as expected.
e.g.
Request-Line: INVITE sip:<B-party>@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: sip:<B-Party>@<SBC-A ip.addr TBD>

Check the wireshark trace and confirm that G.711 A law codec with 10 or 20ms packetisation is being used.
Also check to see if INVITE contains Session-Expires header and that it is syntatically correct. Check for Supported Header to see if 'timer' is supported. Ensure response in 200 OK is compatible with INVITEand check for Required Header and if it contains 'timer'. (x-ref IOP9)
Pass 
IOP5Basic test call from IP-PBX to PSTN line through SBC-B (using SBC-B IPV4 ip address)IP-PBX line initiates call, Call is answered, IP-PBX line terminates call.

Vendors eSBC setup for Solution IP.Addr Mode
Call from the IP-PBX. Invite seen from eSBC to SBC-B, proxy authentication challenge returned to eSBC, re-invite with correct credentials from eSBC and call progresses as expected.
e.g.
Request-Line: INVITE sip:<B-party>@<SBC-B ip.addr TBD>:5060 SIP/2.0
To: sip:<B-Party>@<SBC-B ip.addr TBD>

Check the wireshark trace and confirm that G.711 A law codec with 10ms packetisation is being used.
Pass 
OP7bCalled Number format - vendors eSBC to soft switch number normalisation - Global Dial Plan

Test eSBC capability to send the called number in  one of the following Global number formats (user part of  Request & To URIs)

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

IP-PBX line initiates call to PSTN line, Call is answered.
IP-PBX line terminates call.

Configure the eSBC to present the called number in the user part of the Request & To URIs to be sent in one of the following formats

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass 
IOP8bCalling Number format - vendors eSBC to soft switch number normalisation - Global Dial Plan

Test eSBC capability to send calling number in one of the following Global number formats (user part of FROM & PAI URIs)

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
00yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

IP-PBX line initiates call to PSTN line, Call is answered.
IP-PBX terminates call.

Configure the eSBC to present the calling number in the user part of the From & PAI URIs to be sent in the one of the following formats

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
00yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass 
IOP9bCalled Number format - soft switch to eSBC number normalisation - Global Dial Plan

Test eSBC capability of accepting the called number in one of the following Global number formats (user part of Request & To URIs)

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

PSTN line initiates call to IP-PBX line, Call is answered.
PSTN line terminates call.

Configure the eSBC to accept the called number in the user part of the Request & To URIs in one of the following formats

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)

Also check to see that the INVITE contains Session-Expires header and that it is syntactically correct. Check for Supported Header and ensure 'timer' is supported. Ensure response in 200 OK is compatible with INVITE and check for Required Header and if it contains 'timer'.
Pass 
IOP10bCalling Number format - soft switch to eSBC number normalisation - Global Dial Plan

Test eSBC capability of accepting the calling number in one of the following Global number formats (user part of FROM & PAI URIs)  

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

PSTN line initiates call to IP-PBX line, Call is answered.
PSTN line terminates call.

Configure the eSBC to accept the calling number in the user part of the Request & To URIs in one of the following formats

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass 
IOP11Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 999Call made from IP-PBX line to the Emergency services using 999. Call answered.
Either party terminates call.
e.g.
Request-Line: INVITE sip:999@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:999@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>
Pass 
IOP12Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 112Call made from IP-PBX line to the Emergency services using 112. Call answered,
Either party terminates call.
e.g.
Request-Line: INVITE sip:112@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:112@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>
Pass 
IOP13Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 18000 - Text DirectCall made from IP-PBX line using a text direct set to the Emergency services using 18000. Call answered.
Either party terminates call.
e.g.
Request-Line: INVITE sip:18000@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:18000@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>
Pass_With_Caveat 
IOP14IP-PBX Line to PSTN - call answer - Originator disconnectCall made from IP-PBX line to PSTN line, Answer Call.
IP-PBX line terminates call.
Pass 
IOP15IP-PBX Line to PSTN - call answer - Terminator disconnectCall made from IP-PBX line to PSTN line, Answer Call.
PSTN line terminates call
Pass 
IOP16IP-PBX Line to PSTN - Busy subscriberCall made from IP-PBX line to a busy PSTN line (without divert on busy)
Wait for soft switch to return busy response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.
Pass 
IOP17IP-PBX Line to PSTN - No answer timeout testCall made from IP-PBX line to a PSTN line (without divert on no answer)
Do not answer call.
Wait for soft switch to return no answer timeout response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.
Pass_With_Caveat 
IOP18IP-PBX Line to PSTN - Subscriber not reachableCall made from IP-PBX line to an invalid number.
Wait for soft switch to return response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.
Pass 
IOP19PSTN Line to IP-PBX - call answer - Originator disconnect. Call made from a PSTN line to an IP-PBX line, Answer Call.
Originator disconnects call.
Pass 
IOP20PSTN Line to IP-PBX - call answer - Terminator disconnectCall made from a PSTN line to an IP-PBX line, Answer Call.
IP-PBX line terminates call.
Pass 
IOP21PSTN Line to IP-PBX - busy subscriberCall made from PSTN line to a busy IP-PBX line  (without divert on busy)
Wait for IP-PBX to return busy response.
NoExecSkype server does not support busy line.
IOP22PSTN Line to IP-PBX - No answer timeout test, Invoked by PBXCall made from a PSTN line to an IP-PBX line  (without divert on no answer) Wait for the IP-PBX to return no answer timeout responsePass 
IOP23PSTN Line to IP-PBX - subscriber not reachableCall made from a PSTN line to an invalid number/unprogrammed DDI on the IP-PBX.
Wait for IP-PBX to return response.
Pass 
IOP24Verify CLIP service on IP-PBX line (incoming call from PSTN) Call made from PSTN line to IP-PBX line. PSTN line is set to allow CLI presentation.
Check that CLI is delivered as expected.
Either party terminates call.
Pass 
IOP25Verify CLIR service on IP-PBX line (incoming call from PSTN)Call made from PSTN line to IP-PBX line. PSTN line is set to restrict CLI presentation.
Check that CLI is not delivered as expected.
Either party terminates call.
Pass 
IOP26Verify CLIP service on PSTN line (outgoing call from IP-PBX, From)Ensure number used in From header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX line sends From header containing Calling Line ID (CLI) in the INVITE.

Ensure that the eSBC allows presentation of its CLI using privacy-header (Privacy: none or privacy-header not present)

Ensure that the expected CLI is presented to the PSTN line.
Either party terminates call.
Pass 
IOP27Verify CLIP service on PSTN line (outgoing call from IP-PBX, PAI/PPI)Ensure number used in PAI/PPI header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX line sends PAI/PPI header containing Calling Line ID (CLI) in the INVITE.
If PAI header is populated this will be used in preference to the From header.
Ensure that the eSBC allows presentation of its CLI using privacy-header (Privacy: none or privacy-header not present)

Ensure that the expected CLI is presented to the PSTN line.
Either party terminates call.
FailProblem with PAID header. It is raised with vendor of C20.
IOP28Verify CLIR service on PSTN line (outgoing call from IP-PBX)Ensure number used in From/PAI header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX  line sends From and/or PAI header containing either the Calling Line ID or obscured information in the INVITE.
e.g.
From: "user751000" <sip:+441256751000@192.168.1.10>;tag=12345
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=12345

Ensure that the eSBC restricts presentation of its CLI using privacy-header (Privacy: id or Privacy: user or Privacy: user;id)

Ensure that CLI is NOT presented to the PSTN line.
Either party terminates call.
Pass 
IOP29Verify Call Forward Immediate (unconditional) on a IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)Call made from a PSTN line to an IP-PBX line with call forward to a line within the same IP-PBX, Answer Call.
Either party terminates call.

Does the IP-PBX has configuration settings to send SIP status 181 messages to the soft switch?
Pass 
IOP30Verify Call Forward Immediate (unconditional) on a IP-PBX line (Incoming call from PSTN, call forward terminates PSTN)Call made from a PSTN line to an IP-PBX line with call forward to a line in the PSTN, Answer Call.
Either party terminates call.
Pass 
IOP31Verify Call Forward Busy on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)Call made from a PSTN line to an IP-PBX line with Call Forward Busy (or equivalent) to a line within the IP-PBX, Answer Call.
Either party terminates call.
NoExecSkype server does not support busy line.
IOP32Verify Call Forward No-answer on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)Call made from a PSTN line to an IP-PBX line with Call Forward No-answer (or equivalent) to a line within the IP-PBX, Answer Call.
Either party terminates call.
Pass 
IOP33Verify Call Hold Service on IP-PBX (Incoming call from PSTN)Call made from a PSTN line to an IP-PBX line with Call Hold, Answer call.
IP-PBX line puts the call on hold.
Leave call on hold for 30 seconds and then retrieve call. Ensure speech path is re-established in both directions.
Either party terminates call.
Pass 
IOP34Verify 3-party conference service on IP-PBX (Incoming call from PSTN, 3rd party within IP-PBX)Call made from a PSTN line to an IP-PBX line with 3-party conference, Answer call.
IP-PBX line uses the 3-party conference facility to put PSTN line on hold whilst dialling 3rd party. (another IP-PBX line)
Once the 3rd party has answered the call, place the 3 parties in a conference.
Ensure that all parties have a two way speech path.
Keep the speech path open for at least 20 seconds.
Either party terminates call.
Pass 
IOP35Verify 3-party conference service on IP-PBX (Incoming call from PSTN, 3rd party PSTN)Call made from a PSTN line to an IP-PBX line with 3-party conference, Answer call.
IP-PBX line uses the 3-party conference facility to put PSTN line on hold whilst dialling 3rd party. (another PSTN line)
Once the 3rd party has answered the call, place the 3 parties in a conference.
Ensure that all parties have a two way speech path.
Keep the speech path open for at least 20 seconds.
Either party terminates call.
Pass 
IOP36Verify do-not-disturb service on IP-PBX line (Incoming call from PSTN)Call made from a PSTN line to an IP-PBX line with do-not-disturb feature active. Ensure IP-PBX line does not ring
PSTN line receives an appropriate announcement or tone

Record the SIP status received from IP-PBX
Pass 
IOP37Verify Call park service on IP-PBX line (Incoming call from PSTN)Call made from a PSTN line to IP-PBX line A with Call Park (or equivalent) feature active, Answer call.
Place the call in the Park condition.
After 10 seconds, retrieve call from IP-PBX line B using the Call Park pick-up code.
Ensure speech path is re-established in both directions.
Either party terminates call.
Pass 
IOP38Verify Call Waiting on an IP-PBX line, involving a PSTN lineCall made from PSTN line A to an IP-PBX line with Call Waiting active, Answer call.
Call made from PSTN line B to the same IP-PBX line which should receive an indication that a second call is waiting.
PSTN line B receives ringback tone.
IP-PBX line answers the call from PSTN line B.
PSTN line A should receive an appropriate indication that they are now on hold.
IP-PBX line toggles the call back to PSTN line A
Ensure speech path is re-established in both directions and that PSTN line B should receive an appropriate indication that they are now on hold.
Either party terminates call.
Pass 
IOP39Verify DTMF transmission from/to IP-PBX - InbandConfigure the IP-PBX/eSBC to send DTMF transmission in-band.

Call made from IP-PBX line to a PSTN line, Answer call.
PSTN line presses each of the keys on the number pad in turn. Note the far end experience.
IP-PBX line presses each of the keys on the number pad in turn. Note the far end experience.

Was the received DTMF tone reflective the length of time the key was pressed?
Pass 
IOP40Verify DTMF transmission from/to IP-PBX - RFC 2833 - telephone-event Configure the IP-PBX/eSBC to send DTMF transmission using RFC 2833 - telephone-event.

Call made from IP-PBX line to a PSTN line, Answer call.
PSTN line presses each of the keys on the number pad in turn. Note the far end experience.
IP-PBX line presses each of the keys on the number pad in turn. Note the far end experience.

Was the received DTMF tone reflective the length of time the key was pressed?
Pass 
IOP41T.38 Fax transmission mode - PSTN to IP-PBX originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using T.38 Version 0 Fax transmission mode.
Call made from PSTN line to an IP-PBX line, Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that T.38 Fax Transmission is used. Check that the fax is transmitted and received as expected.
Pass 
IOP42T.38 Fax transmission mode - IP-PBX to PSTN originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using T.38 Version 0 Fax transmission mode.
Call made from IP-PBX line to a PSTN line Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that T.38 Fax Transmission is used. Check that the fax is transmitted and received as expected.
Pass_With_Caveat 
IOP43In-band G.711 Fax transmission mode - PSTN to IP-PBX originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using in-band G.711 Fax transmission mode.
Call made from PSTN line to an IP-PBX line, Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that  in-band G.711 Fax Transmission is used. Check that the fax is transmitted and received as expected.
Pass 
IOP44In-band G.711 Fax transmission mode - IP-PBX to PSTN originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using  in-band G.711 Fax transmission mode.
Call made from IP-PBX line to a PSTN line, Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that  in-band G.711  Fax Transmission is used. Check that the fax is transmitted and received as expected.
Pass 
IOP45Test of Call in progress audit function - response to in-call OPTIONS from soft switch to eSBC.Call made from an IP-PBX line to a PSTN line, Answer call.
Leave the  two parties in conversation for 10 minutes.
Ensure both parties have two way speech.
Either party terminates call.

Check wireshark trace to ensure that in-call OPTIONS are sent by the soft switch and that the eSBC responds with status 200OK. Check to see if the eSBC sends any in-call audit SIP messages.
Pass 
IOP46Test of 4 simultaneous calls, 2 inbound, 2 outbound callsConfigure the eSBC such that successive calls route to alternate SBCs (round robin, cyclic etc).
Make 4 simultaneous calls 2 inbound, 2 outbound calls. Answer calls and ensure two way speech path for each call. 
Pass 
IOP47Test of eSBC endpoint restart-recoveryRestart the eSBC and ensure that, after recovery, inbound and outbound calls are successful.Pass 
IOP48Test of eSBC loss of Ethernet link and reconnectionRemove the Ethernet link between the eSBC and CE router. Leave in this condition for at least 3 minutes. Reconnect the Ethernet link and ensure that after approx 2 minutes inbound and outbound calls are successful.Pass 
IOP49Test of Primary SBC loss ** Contact MSL engineer to carry out the following **
On the Primary SBC carry out the ALLSTOP command to disable the SBC.

Call made from IP-PBX line to a PSTN Line.
Call should attempt to route to Primary SBC. On non-response to INVITE, eSBC re-routes the call to the Secondary SBC.
Wait for call answer.
Either party terminates call.

** Contact MSL engineer to carry out the following **
Restart the Primary SBC
Pass 
IOP50Test of eSBC response to UPDATE messages** Contact MSL engineer to carry out the following **
Run UPDATE emulator script ensuring emulator line points to primary SBC

Call made from IP-PBX line to emulator line as provided by MSL engineer.
eSBC should send a packetization time of 10ms
Pass_With_Caveat 

 

Conclusion

This Application Note describes the steps required to configure the Sonus SBC 1000/2000 to successfully interoperate with Skype for Business 2015 and Virgin Media SIP Trunk. All feature and serviceability test cases are complete. Majority of test cases passed with noted exceptions and observations provided in Interoperability Test Results.