Ribbon EdgeMarc 6000 configuration Microsoft Teams

Public IP address of NAT device (must be Static)*

Private IP address of the SBC

Valid names:

 rbbnvoice.com

Valid names:

• sbc2.rbbnvoice.com
• emea.rbbnvoice.com
• apac.rbbnvoice.com

Invalid name:

• sbc2.emea.rbbnvoice.com

Vendors eSBC response to SIP OPTIONS messages from SBC

No calls are required for this test.

Capture the SIP trace for approximately 60 seconds and check for correct signaling.

For each eSBC, the SBC periodically sends an OPTIONS request to the vendors eSBC to check if its SIP stack is reachable. If the IP-PBX sends a SIP response 200 OK, the SIP trunk is placed or remains in an In-Service state.

Example:

OPTIONS sip:ping@<ip-pbx_IP_Addr>:5060 SIP/2.0

No calls are required for this test.

Capture SIP trace for approximately 60 seconds (depending on agreement) and check for correct signaling.

Vendors eSBC setup for Solution IP.Addr Mode

• The eSBC is configured to send OPTIONS messages to the SBC periodically.
• The SBC responds with SIP response 200 OK.

Example:
"OPTIONS sip:ping@192.168.1.10:5060 SIP/2.0"

• Verify that the eSBC can simultaneously send SIP OPTIONS messages to both the solution SBC addresses.

• The IP-PBX line initiates the call. 
• The call is answered.
• The IP-PBX line terminates the call.

Vendors eSBC setup for Solution IP.Addr Mode

A call progresses successfully when:

• A call is received from the IP-PBX.
• An Invite is seen from the eSBC to SBC-A.
• Proxy authentication challenge is returned to the eSBC.
• A Re-invite with correct credentials is received from the eSBC. 

Example:
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 to confirm that the G.711 A law codec with 10 or 20ms packetization is used.
• Verify that the INVITE contains the Session-Expires header, and the INVITE is syntactically correct.
• Check the Supported Header to ensure that it supports the 'timer'. Ensure that the response in the 200 OK is compatible with the INVITE. Also, verify that the Required Header contains the 'timer'. 

Basic test call from IP-PBX to PSTN line through SBC-B (using SBC-B IPV4 ip address)

Vendor to configure eSBC so that it used secondary SBC (SBC_B) for this test

Once the test completes, eSBC to be configured to use Primary SBC-A for calls to route to

• The IP-PBX line initiates the call. 
• The call is answered.
• The IP-PBX line terminates the call.

Vendors eSBC setup for Solution IP.Addr Mode

A call progresses successfully when:

• A call is received from the IP-PBX.
• An Invite is seen from the eSBC to SBC-B.
• Proxy authentication challenge is returned to the eSBC.
• A re-invite with correct credentials is received from the eSBC.

Example:
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 to confirm that the G.711 A law codec with 10ms or 20ms packetization is used.

Called Number format - vendors eSBC to soft switch number normalization - 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)

Configure the SBC for Global calling plan.

• The IP-PBX line initiates a call to the PSTN line.
• The call is answered.
• The IP-PBX line terminates the call.
• Configure the eSBC to present the called number in the user part of the Request & To URIs and send it 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)

Calling Number format - vendors eSBC to soft switch number normalization - 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)

Configure the SBC for Global calling plan.

• The IP-PBX line initiates a call to the PSTN line.
• The call is answered.
• The IP-PBX terminates the call.
• Configure the eSBC to present the calling number in the user part of the From & PAI URIs and send it 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)
  • 00yyyyyyyyyy (where y refers to any number, calling party = international)
  • yyyyyyyyyy (where y refers to any number, calling party = unknown)

Called Number format - soft switch to eSBC number normalization - 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)

Configure the SBC for Global calling plan.

• The PSTN line initiates a call to the IP-PBX line. 
• The call is answered.
• The PSTN line terminates the 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)
• Verify that the INVITE contains the Session-Expires header and the INVITE is syntactically correct.
• Check the Supported Header to ensure that it supports the 'timer'. Ensure that the response in the 200 OK is compatible with the INVITE. Also, verify that the Required Header contains the 'timer'.

Calling Number format - soft switch to eSBC number normalization - 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)

Configure the SBC for Global calling plan.

• The PSTN line initiates a call to the IP-PBX line.
• The call is answered.
• The PSTN line terminates the 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)

• Make a call from the IP-PBX line to the Emergency services using 999.
• The call is answered.
• Either party terminates the call.

Example:
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>

Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 112

• Make a call from the IP-PBX line to the Emergency services using 112.
• The call is answered.
• Either party terminates the call.

Example:
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>

Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 18000 - Text Direct

• Make a call from the IP-PBX line using a text direct set to the Emergency services using 18000. 
• The call is answered.
• Either party terminates the call.

Example:
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>

• Make a call from the IP-PBX line to the PSTN line.
• Answer the call.
• The IP-PBX line terminates the call.

• Make a call from the IP-PBX line to the PSTN line.
• Answer the call.
• The PSTN line terminates the call.

• Make a call from the IP-PBX line to a busy PSTN line (without divert on busy).
• Wait for the soft switch to return the busy response.
• Ensure that the eSBC is not recursive.
• Set up the call via the secondary SIP trunk.

• Make a call from the IP-PBX line to a PSTN line (without divert on no answer).
• Do not answer the call.
• Wait for the soft switch to return the no answer timeout response.
• Ensure that the eSBC is not recursive. 
• Set up the call via the secondary SIP trunk.

IP-PBX Line to PSTN - Subscriber not reachable

Vendor to call 01189111111

• Make a call from the IP-PBX line to an invalid number.
• Wait for the soft switch to return a response.
• Ensure that the eSBC is not recursive. 
• Set up the call via the secondary SIP trunk.

• Make a call from a PSTN line to an IP-PBX line.
• Answer the call. 
• The originator disconnects the call.

• Make a call from a PSTN line to an IP-PBX line.
• Answer the call.
• The IP-PBX line terminates call.

• Make a call from a PSTN line to a busy IP-PBX line (without divert on busy).
• Wait for the IP-PBX to return the busy response.

• Make a call from a PSTN line to an IP-PBX line (without divert on no answer).
• Wait for the IP-PBX to return the no answer timeout response.

• Make a call from a PSTN line to an invalid number/unprogrammed DDI on the IP-PBX.
• Wait for the IP-PBX to return a response.

• Make a call from the PSTN line to the IP-PBX line.
• The PSTN line is set to allow the CLI presentation. Check that the CLI is delivered as expected.
• Either party terminates call.

• Make a call from the PSTN line to the IP-PBX line.
• PSTN line is set to restrict the CLI presentation. Check that CLI is not delivered as expected.
• Either party terminates call.

• Ensure the number used in From header is agreed with the Virgin Media and entered into the soft switch database for screening.
• Make a call from an IP-PBX line to a PSTN line.
• Ensure the eSBC configuration enables the IP-PBX line to send the From header containing the Calling Line ID (CLI) in the INVITE.
• Ensure that the eSBC allows the presentation of its CLI, using the privacy-header (Privacy: none or privacy-header not present).
• Ensure that the expected CLI is presented to the PSTN line.
• Either party terminates call.

Verify CLIP service on PSTN line (outgoing call from IP-PBX, PAI/PPI)

Vendor to ensure PAI number is different to that from which the call originates

• Ensure the number used in the PAI/PPI header is agreed with the Virgin Media and entered into the soft switch database for screening.
• Make a call from an IP-PBX line to a PSTN line.
• Ensure that the eSBC configuration enables the IP-PBX line to send the PAI/PPI header containing the Calling Line ID (CLI) in the INVITE. Note that if the PAI header is populated, it will be used in preference to the From header.
• Ensure that the eSBC allows the presentation of its CLI, using the privacy-header (Privacy: none or privacy-header not present).
• Ensure that the expected CLI is presented to the PSTN line.
• Either party terminates call.

• Ensure the number used in the From/PAI header is agreed with the Virgin Media and entered into the soft switch database for screening.
• Make a call from an IP-PBX line to a PSTN line. 
• Ensure that the eSBC configuration enables the IP-PBX line to send the From and/or PAI header, containing either the Calling Line ID or obscured information in the INVITE.

Example:
From: "user751000" <sip:+441256751000@192.168.1.10>;tag=12345
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=12345

• Ensure that the eSBC restricts the presentation of its CLI, using the 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.

• Make a call from a PSTN line to an IP-PBX line with Call Forward to a line within the same IP-PBX.
• Answer the call.
• Either party terminates call.

The IP-PBX does not have configuration settings to send SIP status 181 messages to the soft switch.

• Make a call from a PSTN line to an IP-PBX line with Call Forward to a line in the PSTN.
• Answer the call.
• Either party terminates call.

• Make a call from a PSTN line to an IP-PBX line with Call Forward Busy (or equivalent) to a line within the IP-PBX.
• Answer the call.
• Either party terminates call.

• Make a call from a PSTN line to an IP-PBX line with Call Forward No-answer (or equivalent) to a line within the IP-PBX.
• Answer the call.
• Either party terminates call.

• Make a call from a PSTN line to an IP-PBX line with Call Hold.
• Answer the call.
• IP-PBX line places the call on hold.
• Leave the call on hold for 30 seconds and then retrieve call.
• Ensure speech path is re-established in both directions.
• Either party terminates call.

• Make a call from a PSTN line to an IP-PBX line with a 3-party conference.
• Answer the call.
• IP-PBX line uses the 3-party conference facility to place the PSTN line on hold while dialing the 3rd party (on another IP-PBX line).
• Once the 3rd party answers 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.

• Make a call from a PSTN line to an IP-PBX line with a 3-party conference.
• Answer the call.
• IP-PBX line uses the 3-party conference facility to place the PSTN line on hold while dialing the 3rd party (on 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.

• The call does not ring.
• PSTN line receives an appropriate announcement or tone.
• Record the SIP status received from the IP-PBX.

• Make a call from a PSTN line to IP-PBX line A with Call Park (or equivalent) feature active.
• Answer the call.
• Place the call in the Park condition.
• After 10 seconds, retrieve the call from the IP-PBX line B, using the Call Park pick-up code.
• Ensure the speech path is re-established in both directions.
• Either party terminates call.

• Make a call from PSTN line A to an IP-PBX line with Call Waiting active.
• Answer the call. 
• Make a call from the PSTN line B to the same IP-PBX line, which should receive an indication that a second call is waiting.
• PSTN line B receives the ringback tone.
• IP-PBX line answers the call from the 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 the speech path is re-established in both directions and that PSTN line B receives an appropriate indication that they are now on hold.
• Either party terminates call.

• Configure the IP-PBX/eSBC to send the DTMF transmission in-band.
• Make a call from IP-PBX line to a PSTN line.
• Answer the 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.

The received DTMF tone is reflective of the length of time the key was pressed.

• Configure the IP-PBX/eSBC to send the DTMF transmission, using RFC 2833 - telephone-event.
• Make a call from IP-PBX line to a PSTN line.
• Answer the 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.

The received DTMF tone is reflective the length of time the key was pressed.

• Configure the ATA/IP-PBX/eSBC, so that the Fax transmission is sent using the T.38 Version 0 Fax transmission mode.
• Make a call from PSTN line to an IP-PBX line.
• Answer the call. 
• Fax transmission is completed, and the call is terminated by either of the end terminal devices.
• Ensure the Wireshark trace shows it is using the T.38 Fax Transmission. Check that the fax is transmitted and received as expected.

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

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

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

• Make a call from an IP-PBX line to a PSTN line.
• Answer the call.
• Leave the two parties in conversation for 35 minutes.
• Ensure the Session-expires setting is 3600 or less.
• Ensure both parties have two-way speech at the beginning and end of call.
• Either party terminates the call.
• Check the Wireshark trace to ensure that the in-call OPTIONS are sent by the soft switch, and that the eSBC responds with status 200OK.
• Check if the eSBC sends any in-call audit SIP messages.
• Check for session refresh Update or Re-Invite and correct response.

Test for 4 simultaneous calls: 2 inbound, 2 outbound calls

Vendor to configure eSBC for Round robin to ensure calls go to both Primary and secondary SBC

• Configure the eSBC, so that successive calls route to alternate SBCs (round robin, cyclic, and so on).
• Make 4 simultaneous calls: 2 inbound and 2 outbound calls.
• Answer the calls and ensure two-way speech path for each call.

• Restart the eSBC.
• Ensure that, after recovery, inbound and outbound calls are successful.

• Remove 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 approximately 2 minutes inbound and outbound calls are successful.

** Contact MSL engineer to carry out the following **

• On the Primary SBC, carry out the ALLSTOP command to disable the SBC.
• Make a call from the IP-PBX line to a PSTN Line.
• Make sure that the call tries to route to the Primary SBC. On a non-response to INVITE, the 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.

Additional test to cover when vendors are using Microsoft Skype for Business 2015:

• PBX Subscriber 1 makes a call to PBX Subscriber 2, so that the PSTN to call the PBX subscriber 1 is Busy.
• PSTN calls PBX user 1. The call should automatically go to voicemail after 10 seconds when call forwarding is off.
• If VM is on another PBX Internal Line, the call should go to voicemail.
• PSTN user listens to the voiceMail announcement, and leaves a clear message for PBX Subscriber 1 in VM.
• If forwarded to voicemail, the PSTN terminates the call after hearing the VM announcement.
• If forwarded to another user, either party terminates the call after checking that speech is clear in both directions.

Additional test to cover when vendors are using Microsoft Skype for Business 2015:

• PSTN calls PBX user 1.
• The PBX User 1 should not to answer the call.
• The call should automatically go to voicemail (VM), which is in another internal PBX line if call forwarding is turned off.
• The call automatically goes to voicemail after 10 seconds.
• The PSTN terminates the call after hearing the VM announcement.
• If call forwarding is ON, the call is forwarded to another PBX user internal.
• Check the speech quality, and terminate the call after checking that speech is clear in both directions.

• Configure the eSBC to offer the T.38 in addition to G711A-law and G711-U law.
• Make a call from the PBX to a PSTN.
• Ensure the call is connected and dialog takes place for 10 minutes.
• Check the Wireshark output. Confirm that the T.38 is not reflected in the protocol column after the call is connected for 7 minutes.
• If T.38 is reflected in the protocol column, note it down.