Howto:Vozelia 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}}
'''Innovaphone Compatibility Test Report'''
'''Innovaphone Compatibility Test Report'''


== Summary ==
== Summary ==


'''SIP Provider: Name'''
'''SIP Provider: Vozelia'''


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]].  


The provider operates both in France and Spain.


...
Clip No Screening is not supported. Also T.38 and T.38 Transcoding are not supported, so we can't use Innovaphone FaxServer with this SIP Provider.


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]]
That being said, the provider has achieved 87% of all possible test points(138/157). For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#|Test Description]]
 
<!-- in case of customer testimonial, please make sure that the fact that it is not tested by innovaphone is also mentioned in the summary-->


* Features:
* Features:


** Direct Dial In
** Direct Dial In
** Fax over IP (T.38)
** DTMF
** DTMF


Line 25: Line 21:
** G711
** G711
** G729
** G729
** G723
** G726
** T.38 UDP


== Current test state ==
== Current test state ==
Line 35: Line 28:
<!-- {{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=Vozelia_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 April 15th, 2014.
 
<internal>
Beim Abschluss des Tests (egal ob gut, schlecht oder abgebrochen) <strong>bitte Nachricht an ckl</strong>!
</internal>


== Testing Enviroment ==
== Testing Enviroment ==
Line 51: Line 40:


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 without Media Relay and without exclusive coder. This is the case if all tests were successful
* 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 ==


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 ===
Line 73: Line 54:
|----
|----
|'''call using g711a'''
|'''call using g711a'''
|
|'''OK'''
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|
|'''OK'''
|----
|----
|call using g723
|call using g723
|
|NOK
|----
|----
|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
|----
|T.38 Transcoding by the provider
|NOK
|----
|----
|Reverse Media Negotiation
|Reverse Media Negotiation
|
|OK
|----
|----
|CGPN can be suppressed
|CGPN can be suppressed
|
|OK
|----
|----
|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
|
|OK
|----
|----
|Redundancy
|Redundancy
|
|OK
|----
|----
|SIP over TCP
|SIP over TCP
|
|NOK
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|
|'''OK'''
|}
|}


Line 131: Line 115:
|----
|----
|'''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'''
|}
|}


Line 147: Line 131:
|----
|----
|'''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'''
|}
|}


Line 163: Line 147:
|----
|----
|'''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
|}
|}


Line 176: Line 160:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''OK'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|
|OK
|}
|}


Line 190: Line 174:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|
|OK
|}
|}


Line 221: Line 205:
|----
|----
|'''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'''
|}
|}


Line 239: Line 223:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|
|OK
|}
|}


Line 270: Line 254:
|----
|----
|Call can be transferred
|Call can be transferred
|
|OK
|----
|----
|Held end hears dialling tone
|Held end hears dialling tone
|
|OK
|}
|}


Line 283: Line 267:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|}
|}


Line 308: Line 292:
|----
|----
|'''Call can be forward'''
|'''Call can be forward'''
|
|'''OK'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''OK'''
|}
|}


Line 321: Line 305:
|----
|----
|'''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 334: Line 318:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|}
|}


Line 353: Line 337:
|----
|----
|'''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 366: Line 350:
===Firmware version===
===Firmware version===


All innovaphone devices use Vx build xx-xxxxx as firmware.
All innovaphone devices use V10 Sr8


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


[[Image:Vozelia_SIP_Provider_Compatibility_Test_1.png]]


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


[[Image:Vozelia_SIP_Provider_Compatibility_Test_2.png]]
* All incoming calls come in e164 format.


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


[[Image:Vozelia_SIP_Provider_Compatibility_Test_3.png]]
Note: It's necessary to enable Force Enblock in the outgoing route.


=== Media Relay ===
=== Redundancy ===


* When registering the same account on 2 different devices (PBX and Standby PBX) the provider will send all incoming calls to the last device that "registered" on the SIP Server. Since both PBX and Standby PBX have Gateway SIP Interface registered incoming calls can be accepted on both devices at anytime (not simultaneous) so it's not a problem if the SIP Provider sends sometimes the call to Standby-PBX and other times to the Main PBX.
* For outgoing calls the SIP Provider accept calls from both devices/registrations.


=== Fax ===




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

Latest revision as of 14:12, 10 June 2014

Innovaphone Compatibility Test Report

Summary

SIP Provider: Vozelia

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

The provider operates both in France and Spain.

Clip No Screening is not supported. Also T.38 and T.38 Transcoding are not supported, so we can't use Innovaphone FaxServer with this SIP Provider.

That being said, the provider has achieved 87% of all possible test points(138/157). For more information on the test rating, please refer to Test Description

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

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 April 15th, 2014.

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 and without exclusive coder. This is the case if all tests were successful

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 NOK
call using g729 OK
call using g722 NOK
Overlapped sending NOK
early media channel OK
Fax using T.38 NOK
T.38 Transcoding by the provider NOK
Reverse Media Negotiation OK
CGPN can be suppressed OK
CLIP no screening NOK
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection OK
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 PSTN-phone. OK OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. 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 PSTN-phone. OK OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. 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 PSTN-phone. OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. 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 PSTN-phone. OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. 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 V10 Sr8

SIP - Trunk

Vozelia SIP Provider Compatibility Test 1.png

Number Mapping

Vozelia SIP Provider Compatibility Test 2.png

  • All incoming calls come in e164 format.

Route Settings

Vozelia SIP Provider Compatibility Test 3.png

Note: It's necessary to enable Force Enblock in the outgoing route.

Redundancy

  • When registering the same account on 2 different devices (PBX and Standby PBX) the provider will send all incoming calls to the last device that "registered" on the SIP Server. Since both PBX and Standby PBX have Gateway SIP Interface registered incoming calls can be accepted on both devices at anytime (not simultaneous) so it's not a problem if the SIP Provider sends sometimes the call to Standby-PBX and other times to the Main PBX.
  • For outgoing calls the SIP Provider accept calls from both devices/registrations.
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