Howto:Perustele Oy SIP Provider Compatibility Test: Difference between revisions
m (→Known Issues) |
|||
(45 intermediate revisions by the same user not shown) | |||
Line 1: | Line 1: | ||
{{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''' | ||
Line 5: | Line 5: | ||
== Summary == | == Summary == | ||
'''SIP Provider: | '''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. | |||
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--> | ||
Line 19: | Line 20: | ||
** Direct Dial In | ** Direct Dial In | ||
** DTMF | ** DTMF | ||
Line 27: | Line 27: | ||
** G723 | ** G723 | ||
** G726 | ** G726 | ||
== Current test state == | == Current test state == | ||
Line 35: | Line 34: | ||
<!-- {{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=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 March 4th, 2012. | |||
== Testing Enviroment == | == Testing Enviroment == | ||
Line 54: | Line 53: | ||
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. | ||
=== Basic Call === | === Basic Call === | ||
Line 63: | Line 61: | ||
|---- | |---- | ||
|'''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''' | ||
|} | |} | ||
Line 127: | Line 125: | ||
|---- | |---- | ||
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)''' | |'''Loop In call(Innovaphone -> Provider -> Innovaphone)''' | ||
| | |'''OK''' | ||
|} | |} | ||
Line 137: | Line 135: | ||
|---- | |---- | ||
|'''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 153: | Line 151: | ||
|---- | |---- | ||
|'''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 166: | Line 164: | ||
|---- | |---- | ||
|'''Call can be transferred''' | |'''Call can be transferred''' | ||
| | |'''OK''' | ||
|---- | |---- | ||
|Held end hears music on hold | |Held end hears music on hold | ||
| | |OK | ||
|} | |} | ||
Line 180: | Line 178: | ||
|---- | |---- | ||
|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 | ||
|} | |} | ||
Line 207: | 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 225: | Line 223: | ||
|---- | |---- | ||
|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 | ||
|} | |} | ||
Line 252: | Line 250: | ||
|---- | |---- | ||
|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. | ||
Line 265: | Line 265: | ||
|---- | |---- | ||
|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 | ||
|} | |} | ||
Line 287: | Line 287: | ||
|---- | |---- | ||
|'''Call can be forward''' | |'''Call can be forward''' | ||
| | |'''OK''' | ||
|---- | |---- | ||
|'''Held end hears dialling tone''' | |'''Held end hears dialling tone''' | ||
| | |'''OK''' | ||
|} | |} | ||
Line 300: | 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 313: | 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 329: | 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 342: | Line 342: | ||
===Firmware version=== | ===Firmware version=== | ||
All innovaphone devices use | 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 === | |||
* 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. | |||
[[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
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
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
Number Mapping
Route Settings
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.