Howto:E-Fon - ADAPT - 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: E-Fon'''


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]].  


All tests were made without use of Media-Relay or Exclusive Coder.The NAT Detection mechanism and SIP Reverse Media Negotiation work without problems. The SIP Provider supports the G722 codec and does transcoding, by this it allows call to PSTN destinations using G722.


...
Redundancy is not possible since the SIP Provider only allows one active registration per account at a single time. If we register a second device, it will replace the previous registration.


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]]
T.38 is not supported as codec. G711 Pass-through is supported by the provider to send and receive fax calls.
<!-- Mention all important tests that were not passed in the summary.  


E.g. in case that the provider doesn't support Reverse Media-Negotiation, mention in the summary that media relay and an exclusive coder setting must be configured:  
CGPN Suppression is possible, it requires to send as prefix "*31" in the CDPN. This can be automated by an additional [[Howto:E-Fon_-_ADAPT_-_Provider_Compatibility_Test#Route_Settings | Route Table entry]].


Since the provider doesn't support Reverse Media negotiation, media relay and an exclusive coder setting must be configured. Opposed to a SIP trunk not needing Media-Relay, the transport of all RTP packets by the gateway will result in a higher CPU load for a call. As a result, the amount of concurrent calls is considerably lower compared to a SIP-Provider that doesn't require Media-Relay.
That being said, the provider has achieved 89% of all possible test points (140/157). 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)
** Fax over IP (G711 Pass-through)
** DTMF
** DTMF
** CGPN Suppression
** CLIP No Screening
** NAT Detection
** Reverse Media Negotiation


* Supported Codecs by the provider
* Supported Codecs by the provider
** G711
** G711 a/u
** G729
** G729
** G723
** G722
** G726
** G726 (Not tested but listed on the Media Capabilities)
** T.38 UDP


== Current test state ==
== Current test state ==
Line 41: Line 39:
<!-- {{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=EFON_ADAPT_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 June 23th, 2014.
 
<internal>
Beim Abschluss des Tests (egal ob gut, schlecht oder abgebrochen) <strong>bitte Nachricht an ckl</strong>!
</internal>


== Testing Enviroment ==
== Testing Enviroment ==
Line 57: Line 51:


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 ==
Line 70: Line 58:
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 79: Line 66:
|----
|----
|'''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
|
|OK
|----
|----
|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
|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
|
|OK
|----
|----
|'''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
|
|NOK
|----
|----
|SIP over TCP
|SIP over TCP
|
|NOK
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|
|'''OK'''
|}
|}


Line 140: Line 127:
|----
|----
|'''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 156: Line 143:
|----
|----
|'''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 172: Line 159:
|----
|----
|'''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 185: Line 172:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''OK'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|
|OK
|}
|}


Line 199: Line 186:
|----
|----
|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 230: Line 217:
|----
|----
|'''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 248: Line 235:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|
|NOK
|----
|----
|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 279: Line 266:
|----
|----
|Call can be transferred
|Call can be transferred
|
|OK
|----
|----
|Held end hears dialling tone
|Held end hears dialling tone
|
|OK
|}
|}


Line 292: Line 279:
|----
|----
|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 317: Line 304:
|----
|----
|'''Call can be forward'''
|'''Call can be forward'''
|
|'''OK'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''OK'''
|}
|}


Line 330: Line 317:
|----
|----
|'''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 343: Line 330:
|----
|----
|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 362: Line 349:
|----
|----
|'''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 375: Line 362:
===Firmware version===
===Firmware version===


All innovaphone devices use Vx build xx-xxxxx as firmware.
All innovaphone devices use V10sr11 as firmware.


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


[[Image:EFON_-_SIP_Provider_Compatibility_Test_1.png]]


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


[[Image:EFON_-_SIP_Provider_Compatibility_Test_2.png]]


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


[[Image:EFON_-_SIP_Provider_Compatibility_Test_3.png]]
* Force Enblock is required.
* Special Route to handle Restricted Calls.


=== Media Relay ===
=== Known Limitations ===


* No T.38 since it's rejected with "SIP 488 Not Acceptable Here".


=== Fax ===
* SIP Privacy ID for CGPN Number suppression not supported, as solution we have to do a route that checks the CGPN for number suppresion (R!) and add as prefix to the CDPN the code *31.


* Multiple registrations with the same account are not possible, a new registration replaces the previous one making not possible to do redundancy with 2 gateways.


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

Latest revision as of 14:02, 3 July 2014

Innovaphone Compatibility Test Report

Summary

SIP Provider: E-Fon

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

All tests were made without use of Media-Relay or Exclusive Coder.The NAT Detection mechanism and SIP Reverse Media Negotiation work without problems. The SIP Provider supports the G722 codec and does transcoding, by this it allows call to PSTN destinations using G722.

Redundancy is not possible since the SIP Provider only allows one active registration per account at a single time. If we register a second device, it will replace the previous registration.

T.38 is not supported as codec. G711 Pass-through is supported by the provider to send and receive fax calls.

CGPN Suppression is possible, it requires to send as prefix "*31" in the CDPN. This can be automated by an additional Route Table entry.

That being said, the provider has achieved 89% of all possible test points (140/157). For more information on the test rating, please refer to Test Description

  • Features:
    • Direct Dial In
    • Fax over IP (G711 Pass-through)
    • DTMF
    • CGPN Suppression
    • CLIP No Screening
    • NAT Detection
    • Reverse Media Negotiation
  • Supported Codecs by the provider
    • G711 a/u
    • G729
    • G722
    • G726 (Not tested but listed on the Media Capabilities)

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 June 23th, 2014.

Testing Enviroment

SIPProviderTestTopology1.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 OK
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 OK
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection OK
Redundancy NOK
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 NOK
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 V10sr11 as firmware.

SIP - Trunk

EFON - SIP Provider Compatibility Test 1.png

Number Mapping

EFON - SIP Provider Compatibility Test 2.png

Route Settings

EFON - SIP Provider Compatibility Test 3.png

  • Force Enblock is required.
  • Special Route to handle Restricted Calls.

Known Limitations

  • No T.38 since it's rejected with "SIP 488 Not Acceptable Here".
  • SIP Privacy ID for CGPN Number suppression not supported, as solution we have to do a route that checks the CGPN for number suppresion (R!) and add as prefix to the CDPN the code *31.
  • Multiple registrations with the same account are not possible, a new registration replaces the previous one making not possible to do redundancy with 2 gateways.
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