Howto:QSC IPfonie extended - SIP Provider Compatibility Test: Difference between revisions

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'''Innovaphone Compatibility Test Report'''
'''Innovaphone Compatibility Test Report'''
{{Template:Compat Status "tested"}}


== Summary ==
== Summary ==


'''SIP Provider: QSC'''
'''SIP Provider: QSC'''
The provider supports all required innovaphone features and is therefore qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|recommended SIP Provider]].
Please note that in order to use [[Howto:How_does_"CLIP_no_screening"_work%3F | Clip no screening]], some [[Howto:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test#Known_Issues|extra configuration]] must be done.
That being said, the provider has achieved 96.2% of all possible test points (128 on 133). For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]]


* Features:
* Features:


** T.38 IP Fax
** Direct Dial In
** Direct Dial In
** DTMF
** DTMF
** CGPN can be suppressed
** CLIP No Screening
** SIP over TCP


* Supported Codecs by the provider
* Supported Codecs by the provider
** G711
** G711a/u
** G722
** G729
** G729
** G723
** G723
** T.38
** T.38 UDP


The provider supports all required innovaphone feature.(i.e. reverse media negotiation, DDI, T.38). The configuration on innovaphone side is simple, since the built-in SIP-GW can be used to connect to the provider.
== Current test state ==


QSC does not support the remote hold feature. When making a blind transfer(using redial key) the remote end will not get a MOH/dialtone from the provider.
<!--{{Template:Compat Status "planned"}} -->
<!-- {{Template:Compat Status "in progress"}} -->
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"engineered_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"rec._prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat Status "customer-testimonial"}} -->


Since only minor problems have arizen during the tests, it is not possible to make a statement on the QSC support quality.
{{Template:Compat Status "certified"|certificate=QSC_-_IPfonie_extended_-_SIP_Provider_-_product-cert.pdf}}
 
Testing of this product has been finalized October 11th, 2012.


== Testing Enviroment ==
== Testing Enviroment ==


[[Image:HFO_SIP_Compatibility_Test_5.PNG]]


=== Scenario NAT ===
This scenario describes a setup where the PBX and phones are in a private network.  
 
[[Image:HFO_SIP_Compatibility_Test_5.PNG]]


This scenario describes a setup where the PBX and phones are in a private network. The IP800 must use a stun server, in order to send correct SIP - messages. The IP800 works as media relay, all RTP - streams go through the PBX.
The SIP trunk is configured without Media Relay, no STUN and without exclusive coder.


== Test Results ==
== Test Results ==


For more information on the test procedure, please read the following wiki article: [[Howto:SIP_Interop_Test_Description|SIP Interop Test Description]]. Bold lines in the test results indicate a KO-criteria.


=== Basic Call ===
=== Basic Call ===
Line 40: Line 58:
{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!Result  
|----
|----
|call using g711a
|'''call using g711a'''
|Yes
|'''OK'''
|----
|----
|call using g711u
|'''call using g711u'''
|Yes
|'''OK'''
|----
|----
|call using g723
|call using g723
|Yes
|OK
|----
|----
|call using g729
|call using g729
|Yes
|OK
|----
|call using g722
|OK*
|----
|----
|Overlapped sending
|Overlapped sending
|No
|NOK
|----
|----
|early media channel
|'''early media channel'''
|Yes
|'''OK'''
|----
|----
|Fax using T.38
|Fax using T.38
|Yes
|OK
|----
|----
|CGPN Can be supressed
|Reverse Media Negotiation
|No
|OK
|----
|----
|Reverse media negotiation
|CGPN can be suppressed
|Yes
|OK
|----
|----
|Voice Quality OK?
|CLIP no screening
|Yes
|OK
|----
|'''Long time call possible(>30 min)'''
|'''OK'''
|----
|'''External Transfer'''
|'''OK'''
|----
|NAT Detection
|OK
|----
|Redundancy
|OK
|----
|SIP over TCP
|OK
|----
|'''Voice Quality OK?'''
|'''OK'''
|}
|}
"*" - G722 is only supported if the remote endpoint also supports this codec (QSC doesn't do transcoding of the calls).


=== Direct Dial In ===
=== Direct Dial In ===
Line 77: Line 117:
{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!Result  
|----
|----
|Inbound(Provider -> Innovaphone)
|'''Inbound(Provider -> Innovaphone)'''
|Yes
|'''OK'''
|----
|----
|Outbound(Innovaphone -> Provider)
|'''Outbound(Innovaphone -> Provider)'''
|Yes
|'''OK'''
|}
|}


Line 90: Line 130:
{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!Result  
|----
|----
|DTMF tones sent correctly
|'''DTMF tones sent correctly'''
|Yes
|'''OK'''
|----
|----
|DTMF tones received correctly
|DTMF tones sent correctly via SIP-Info
|Yes
|NOK
|----
|----
|DTMF tones audible in both directions
|'''DTMF tones received correctly'''
|Yes
|'''OK'''
|}
|}


Line 106: Line 146:
{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!Result  
|----
|----
|Call can be put on hold  
|'''Call can be put on hold'''
|Yes
|'''OK'''
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|Yes
|OK
|}
 
=== Transfer with consultation ===
 
{| border="1"
!Tested feature
!Result
|----
|----
|Held end hears music on hold / announcement from provider
|'''Call can be transferred'''
|No
|'''OK'''
|----
|----
|Either call can be terminated or be retrieved
|Held end hears music on hold
|Yes
|OK
|}
|}


=== Transfer with consultation ===
The following tests are made to test if call transfer is working.


{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!Voice Ok?
!MoH Ok?
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|OK
|OK
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|OK
|OK
|----
|----
|Call can be transfered
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|Yes
|OK
|OK
|----
|----
|Held end hears music on hold
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|Yes
|OK
|OK
|----
|----
|Call returns to transferring device if the third
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
Endpoint is not available
|OK
|Yes
|OK
|}
|}


Line 142: Line 200:
{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!Result  
|----
|----
|Call can be transfered
|'''Call can be transferred'''
|Yes
|'''OK'''
|----
|----
|Held end hears music on hold or dialing tone
|Held end hears music on hold or dialling tone
|Yes
|OK
|----
|----
|Call returns to transferring device if the third  
|'''Call returns to transferring device if the third'''
Endpoint is not available
'''Endpoint is not available'''
|Yes
|'''OK'''
|}
 
The following tests are made to test if call transfer is working.
 
{| border="1"
!Tested feature
!Voice Ok?
!MoH Ok?
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|OK
|OK
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|OK
|OK
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|OK
|OK
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|OK
|OK
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|OK
|OK
|}
|}


Line 159: Line 245:
{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result S1
!Result  
|----
|----
|Call can be transfered
|Call can be transferred
|Yes
|OK
|----
|----
|Held end hears dialing tone
|Held end hears dialling tone
|No
|OK
|}
|}


=== Broadcast Group & Waiting Queue ===
The following tests are made to test if call transfer is working.


{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!Voice Ok?
|----
|----
|Caller can make a call to a Broadcast Group
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|Yes
|OK
|----
|----
|Caller can make a call to a Waiting Queue
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|Yes
|OK
|----
|----
|Announcement if nobody picks up the call
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|Yes
|OK
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|OK
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|OK
|}
|}


== Configuration ==
=== Blind Transfer (alerting only)===


=== General Information ===
{| border="1"
!Tested feature
!Result
|----
|Call can be transferred
|OK
|----
|Held end hears dialling tone
|OK
|}


'''Basic Provider Infomation: QSC'''
The following tests are made to test if call transfer is working.


{| border="1"
!Tested feature
!Voice Ok?
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|OK
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|OK
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|OK
|}


*described in Mantis Case: 22169
=== Broadcast Group & Waiting Queue ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Caller can make a call to a Broadcast Group'''
|'''OK'''
|----
|'''Caller can make a call to a Waiting Queue'''
|'''OK'''
|----
|'''Announcement if nobody picks up the call'''
|'''OK'''
|}


'''Firmware version'''
== Configuration ==


===Firmware version===


*IP800: 6.00 dvl-sr2 IP800[07-60600.80]
All innovaphone devices use V9 hotfix 16 build 9.061101 as firmware.
*IP22: 6.00 dvl-sr1 IP230[07-60600.58]
*IP200: 6.00 dvl-sr1 IP230[07-60600.58]
*IP230: 6.00 dvl-sr1 IP230[07-60600.58]


=== SIP - Trunk ===
=== SIP - Trunk ===


First of all the SIP Trunk must be configured. Here an example of our QSC - Trunk. If you want that the Gateway should act as Medialrelay, you must exchange the Gatekeeper Address (blue marking) to an private network address; for example 127.0.0.1. QSC awaits in the From Header the complete Calling Party Number(CGPN). The default innovaphone setting is to not send the complete CGPN in  the FROM - Header, but in the Preffered Identity Header. Change the setting ''From Header:'' to ''CGPN in user part of URI''.
Here's the configuration of the SIP gateway interface.
 


[[Image:QSC SIP Compatibility Test 1.PNG]]
[[Image:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test_1.png]]


=== Number Mapping ===
=== Number Mapping ===


The complicated part on this issue is the correct mapping of the outgoing and incoming numbers.
[[Image:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test_2.png]]
 
[[Image:QSC SIP Compatibility Test 2.PNG]]


=== Route Settings ===
=== Route Settings ===


Because QSC, as most SIP - Providers too, doesn't support overlap sending, you must enable the blockwise sending of the phone number. You can do this by enabling ''Force enblock'' in the automatically generated Routes.
[[Image:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test_3.png]]
 
The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay suplementary services, like Hold over the SIP Trunk. If this checkbox is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.
 
[[Image:QSC SIP Compatibility Test 3.PNG]]
 
=== Media Relay ===
 
By enabling Media Relay on the PBX, The IP800 will work as an RTP - Proxy, so all RTP Streams will travel through the IP800. This mode poses a much greater load on the PBX, so the number of concurrent calls will be heavy limited.
 
[[Image:QSC SIP Compatibility Test 4.PNG]]


You may wonder about the usefullness of putting the localhost address as a private network. However you must insert this entry, if the IP800 SIP GW registers at the PBX on localhost interface.
Force Enblock Setting it's required for outgoing calls. QSIG/SIP Interworking setting it's recommended.


=== Fax ===
=== Redundancy ===


The FAX was connected via a IP22 to the IP800. When configuring the IP22 you must keep in mind that you will use the analog interface for fax communication. Thats why the T.38 codec must be enabled.
* QSC allows more than one registration on the same account. Incoming calls will be delivered on the first device that registered on the account. If the first device goes down or looses its registration, then incoming calls are delivered on the 2nd registered device. When the initial device comes back online then calls will delivered again to it.


[[Image:QSC SIP Compatibility Test 5.PNG]]
=== Known Issues ===


Now the PBX and the phones are setup correctly. You should be able to make call in both directions and send and receive fax messages.
* In order to CLIP No Screening work properly we must set option to send SIP Address P-Asserted Identity like below.


http://x.x.x.x/!config add SIP /pai
http://x.x.x.x/!config write
http://x.x.x.x/!config activate


[[Category:Compat|{{PAGENAME}}]]
[[Category:Compat|{{PAGENAME}}]]

Latest revision as of 14:52, 18 June 2014

Innovaphone Compatibility Test Report

Summary

SIP Provider: QSC

The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.

Please note that in order to use Clip no screening, some extra configuration must be done.

That being said, the provider has achieved 96.2% of all possible test points (128 on 133). For more information on the test rating, please refer to Test Description

  • Features:
    • T.38 IP Fax
    • Direct Dial In
    • DTMF
    • CGPN can be suppressed
    • CLIP No Screening
    • SIP over TCP
  • Supported Codecs by the provider
    • G711a/u
    • G722
    • G729
    • G723
    • T.38 UDP

Current test state

Recprod.PNG The tests for this product have been completed and it has been approved as a recommended product (Certification document).

Testing of this product has been finalized October 11th, 2012.

Testing Enviroment

HFO SIP Compatibility Test 5.PNG

This scenario describes a setup where the PBX and phones are in a private network.

The SIP trunk is configured without Media Relay, no STUN and without exclusive coder.

Test Results

For more information on the test procedure, please read the following wiki article: SIP Interop Test Description. Bold lines in the test results indicate a KO-criteria.

Basic Call

Tested feature Result
call using g711a OK
call using g711u OK
call using g723 OK
call using g729 OK
call using g722 OK*
Overlapped sending NOK
early media channel OK
Fax using T.38 OK
Reverse Media Negotiation OK
CGPN can be suppressed OK
CLIP no screening OK
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection OK
Redundancy OK
SIP over TCP OK
Voice Quality OK? OK

"*" - G722 is only supported if the remote endpoint also supports this codec (QSC doesn't do transcoding of the calls).

Direct Dial In

Tested feature Result
Inbound(Provider -> Innovaphone) OK
Outbound(Innovaphone -> Provider) OK

DTMF

Tested feature Result
DTMF tones sent correctly OK
DTMF tones sent correctly via SIP-Info NOK
DTMF tones received correctly OK

Hold/Retrieve

Tested feature Result
Call can be put on hold OK
Held end hears music on hold / announcement from PBX OK

Transfer with consultation

Tested feature Result
Call can be transferred OK
Held end hears music on hold OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone. OK OK
inno1 calls sip-provider-phone. inno1 transfers to inno2. OK OK
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. OK OK
sip-provider-phone calls inno1. inno1 transfers to inno2. OK OK
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. OK OK

Transfer with consultation (alerting only)

Tested feature Result
Call can be transferred OK
Held end hears music on hold or dialling tone OK
Call returns to transferring device if the third

Endpoint is not available

OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone. OK OK
inno1 calls sip-provider-phone. inno1 transfers to inno2. OK OK
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. OK OK
sip-provider-phone calls inno1. inno1 transfers to inno2. OK OK
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. OK OK

Blind Transfer

Tested feature Result
Call can be transferred OK
Held end hears dialling tone OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone. OK
inno1 calls sip-provider-phone. inno1 transfers to inno2. OK
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. OK
sip-provider-phone calls inno1. inno1 transfers to inno2. OK
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. OK

Blind Transfer (alerting only)

Tested feature Result
Call can be transferred OK
Held end hears dialling tone OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone. OK
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. OK
sip-provider-phone calls inno1. inno1 transfers to inno2. OK

Broadcast Group & Waiting Queue

Tested feature Result
Caller can make a call to a Broadcast Group OK
Caller can make a call to a Waiting Queue OK
Announcement if nobody picks up the call OK

Configuration

Firmware version

All innovaphone devices use V9 hotfix 16 build 9.061101 as firmware.

SIP - Trunk

Here's the configuration of the SIP gateway interface.

QSC IPfonie extended - SIP Provider Compatibility Test 1.png

Number Mapping

QSC IPfonie extended - SIP Provider Compatibility Test 2.png

Route Settings

QSC IPfonie extended - SIP Provider Compatibility Test 3.png

Force Enblock Setting it's required for outgoing calls. QSIG/SIP Interworking setting it's recommended.

Redundancy

  • QSC allows more than one registration on the same account. Incoming calls will be delivered on the first device that registered on the account. If the first device goes down or looses its registration, then incoming calls are delivered on the 2nd registered device. When the initial device comes back online then calls will delivered again to it.

Known Issues

  • In order to CLIP No Screening work properly we must set option to send SIP Address P-Asserted Identity like below.
http://x.x.x.x/!config add SIP /pai
http://x.x.x.x/!config write
http://x.x.x.x/!config activate
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