Howto:Perustele Oy SIP Provider Compatibility Test: Difference between revisions

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{{FIXME|reason=Note: Article - Name should be Howto:Productname-SIP-provider-compatibility-test. It is important that the productname is in the article title and not the provider name or country-a provider could have different products or the same product in more countries}}
<!-- {{FIXME|reason=Note: Article - Name should be Howto:Productname-SIP-provider-compatibility-test. It is important that the productname is in the article title and not the provider name or country-a provider could have different products or the same product in more countries}} -->


'''Innovaphone Compatibility Test Report'''
'''Innovaphone Compatibility Test Report'''
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== Summary ==
== Summary ==


'''SIP Provider: Name'''
'''SIP Provider: Perustele Oy'''


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]].  
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]].  


That being said, the provider has achieved 83,90% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#|Test Description]]


...
During the tests we had some issues regarding Media Negotiation, so to work proper we must set Exclusive Codec and Media-Relay ON in the SIP Trunk.


That being said, the provider has achieved x% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#|Test Description]]
More information could be found at [[Howto:Perustele_Oy_SIP_Provider_Compatibility_Test#Known_Issues|Known Issues]]


<!-- in case of customer testimonial, please make sure that the fact that it is not tested by innovaphone is also mentioned in the summary-->  
<!-- in case of customer testimonial, please make sure that the fact that it is not tested by innovaphone is also mentioned in the summary-->  
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** Direct Dial In
** Direct Dial In
** Fax over IP (T.38)
** DTMF
** DTMF


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** G723
** G723
** G726
** G726
** T.38 UDP


== Current test state ==
== Current test state ==
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<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{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_"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_"rec._prod."|certificate=Perustele_Oy_-_SIP_Provider_-_product-cert.pdf}}
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


<!-- Testing of this product has been finalized January 1st, 1970. -->
Testing of this product has been finalized March 4th, 2012.


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


There are 3 major configuration variants of the SIP trunk, which one is used depends on the test results. The variants are:
* the SIP trunk is configured with Media Relay but without exclusive coder. This is the case when the test for "NAT Traversal" fails
 
* the SIP trunk is configured without Media Relay and without exclusive coder. This is the case if all tests were successful
* the SIP trunk is configured with Media Relay and STUN but without exclusive coder. This is the case when the test for "NAT Traversal" fails
* the SIP trunk is configured with Media Relay and exclusive coder. This is the case when the test for "Reverse Media Negotiation" fails
 
The test scenario should describe which SIP trunk configuration is needed.


== Test Results ==
== Test Results ==
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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.
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.


{{FIXME|reason=Note: Possible test results are Ok, Nok, Not tested, N/A(not available). Ok, Nok, N/A are self explanatory. "Not tested" is more a temporary note - if its still in the test after finishing, then the summary should explain why the feature was not tested. Remove this note in the final report}}


=== Basic Call ===
=== Basic Call ===
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|----
|----
|'''call using g711a'''
|'''call using g711a'''
|
|'''OK'''
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|
|'''OK'''
|----
|----
|call using g723
|call using g723
|
|OK
|----
|----
|call using g729
|call using g729
|
|OK
|----
|----
|call using g722
|call using g722
|
|NOK
|----
|----
|Overlapped sending
|Overlapped sending
|
|NOK
|----
|----
|'''early media channel'''
|'''early media channel'''
|
|'''OK'''
|----
|----
|Fax using T.38
|Fax using T.38
|
|NOK
|----
|----
|Reverse Media Negotiation
|Reverse Media Negotiation
|
|NOK
|----
|----
|CGPN can be suppressed
|CGPN can be suppressed
|
|NOK
|----
|----
|CLIP no screening
|CLIP no screening
|
|NOK
|----
|----
|'''Long time call possible(>30 min)'''
|'''Long time call possible(>30 min)'''
|
|'''OK'''
|----
|----
|'''External Transfer'''
|'''External Transfer'''
|
|'''OK'''
|----
|----
|NAT Detection
|NAT Detection
|
|NOK
|----
|----
|Redundancy
|Redundancy
|
|OK
|----
|----
|SIP over TCP
|SIP over TCP
|
|NOK
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|
|'''OK'''
|}
|}


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|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone)'''
|
|'''OK'''
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|
|'''OK'''
|----
|----
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|
|'''OK'''
|}
|}


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|----
|----
|'''DTMF tones sent correctly'''
|'''DTMF tones sent correctly'''
|
|'''OK'''
|----
|----
|DTMF tones sent correctly via SIP-Info
|DTMF tones sent correctly via SIP-Info
|
|NOK
|----
|----
|'''DTMF tones received correctly'''
|'''DTMF tones received correctly'''
|
|'''OK'''
|}
|}


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|----
|----
|'''Call can be put on hold'''  
|'''Call can be put on hold'''  
|
|'''OK'''
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|
|OK
|}
|}


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|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''OK'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|
|OK
|}
|}


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|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|}
|}


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|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''OK'''
|----
|----
|Held end hears music on hold or dialling tone
|Held end hears music on hold or dialling tone
|
|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'''
|
|'''OK'''
|}
|}


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|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|}
|}


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|----
|----
|Call can be transferred
|Call can be transferred
|
|OK*
|----
|----
|Held end hears dialling tone
|Held end hears dialling tone
|
|OK
|}
|}
"*" - Blind Transfer only works if we set exclusive coder on the SIP Interface.


The following tests are made to test if call transfer is working.
The following tests are made to test if call transfer is working.
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|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|----
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|----
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|OK
|}
|}


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|----
|----
|'''Call can be forward'''
|'''Call can be forward'''
|
|'''OK'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''OK'''
|}
|}


Line 306: Line 300:
|----
|----
|'''Call can be transferred or forward'''
|'''Call can be transferred or forward'''
|
|'''OK'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''OK'''
|}
|}


Line 319: Line 313:
|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|}
|}


Line 335: Line 329:
|----
|----
|'''Caller can make a call to a Broadcast Group'''
|'''Caller can make a call to a Broadcast Group'''
|
|'''OK'''
|----
|----
|'''Caller can make a call to a Waiting Queue'''
|'''Caller can make a call to a Waiting Queue'''
|
|'''OK'''
|----
|----
|'''Announcement if nobody picks up the call'''
|'''Announcement if nobody picks up the call'''
|
|'''OK'''
|}
|}


Line 348: Line 342:
===Firmware version===
===Firmware version===


All innovaphone devices use Vx build xx-xxxxx as firmware.
All innovaphone devices use V9 Hotfix20 as firmware.


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


[[Image:Perustele_Oy_SIP_Provider_Compatibility_Test_1.png]]


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


[[Image:Perustele_Oy_SIP_Provider_Compatibility_Test_2.png]]


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


[[Image:Perustele_Oy_SIP_Provider_Compatibility_Test_3.png]]
=== Redundancy ===
The solution for redundancy, it is not straight forward.
As a workaround in cases with forwarding to SIP URI Port can using the
domain name instead of IP. In this case Provider can configure DNS server to send
the requests to slave PBX if the master PBX is not available.
=== Known Issues ===


=== Media Relay ===
* Reverse Media Negotiation - This feature it's not supported, due that fact it's necessary to set Exclusive Codec and Media-Relay options in the SIP Trunk configuration.


* No T.38 - During the SIP negotation T.38 was accepted however the fax transmission fails.


=== Fax ===
* Interworking/QSIG could cause issues, reported by costumer when performing Hold and using Interworking/QSIG flag on the routes the media change offer sent to the Provider was rejected. So the solution should be not use this option as shown in the configuration above.




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

Latest revision as of 11:22, 12 July 2013


Innovaphone Compatibility Test Report

Summary

SIP Provider: Perustele Oy

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

That being said, the provider has achieved 83,90% of all possible test points. For more information on the test rating, please refer to Test Description

During the tests we had some issues regarding Media Negotiation, so to work proper we must set Exclusive Codec and Media-Relay ON in the SIP Trunk.

More information could be found at Known Issues


  • Features:
    • Direct Dial In
    • DTMF
  • Supported Codecs by the provider
    • G711
    • G729
    • G723
    • G726

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 March 4th, 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 with Media Relay but without exclusive coder. This is the case when the test for "NAT Traversal" fails

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

Direct Dial In

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

"*" - Blind Transfer only works if we set exclusive coder on the SIP Interface.

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

CFU / CFB Transfer

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

CFNR / Blind Transfer (alerting only)

Tested feature Result
Call can be transferred or forward 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 Hotfix20 as firmware.

SIP - Trunk

Perustele Oy SIP Provider Compatibility Test 1.png

Number Mapping

Perustele Oy SIP Provider Compatibility Test 2.png

Route Settings

Perustele Oy SIP Provider Compatibility Test 3.png

Redundancy

The solution for redundancy, it is not straight forward.

As a workaround in cases with forwarding to SIP URI Port can using the domain name instead of IP. In this case Provider can configure DNS server to send the requests to slave PBX if the master PBX is not available.

Known Issues

  • Reverse Media Negotiation - This feature it's not supported, due that fact it's necessary to set Exclusive Codec and Media-Relay options in the SIP Trunk configuration.
  • No T.38 - During the SIP negotation T.38 was accepted however the fax transmission fails.
  • Interworking/QSIG could cause issues, reported by costumer when performing Hold and using Interworking/QSIG flag on the routes the media change offer sent to the Provider was rejected. So the solution should be not use this option as shown in the configuration above.
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