Howto:Netia-S.A. - SIP Provider Compatibility Test: Difference between revisions
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'''Netia S.A. innovaphone Compatibility Test Report''' | '''Netia S.A. innovaphone Compatibility Test Report''' | ||
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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 beeing said, the provider has achieved | That beeing said, the provider has achieved 88% of all possible test points (110 on total of 125 Points). For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]] | ||
They are two methods of setting up a SIP trunk with Netia: | |||
* Trunk without Registration - Requires to provide the SIP Provider the Client IP Public address that will make the trunk (SBC or PBX). | |||
* Trunk with Registration - Requires only right credentials (user and password) don't need to know the IP address of client. | |||
We tested the second method. SIP Trunk with Registration. | |||
* Features: | * Features: | ||
** Direct Dial In | ** Direct Dial In | ||
** DTMF | ** DTMF | ||
** NAT Detection | ** NAT Detection | ||
** | ** Number presentation (CLIR) could be suppressed | ||
** Redundancy | |||
* Supported Codecs by the provider | * Supported Codecs by the provider | ||
** G711 | ** G711 a/u | ||
** G729 | ** G729 | ||
Note: T.38 FAX was tested with this provider during the tests. This feature didn't worked as expected so is listed as "Not Supported" however in future fixes could work with Innovaphone probably. | |||
* Redundancy Methods: | |||
A) Using Session Agent Group on SBC - only static SIP Trunk (SIP Trunk | |||
without registration). | |||
B) Using Enterprise Group on AS - using OTG/DTG Identity. | |||
Note: For redundancy Netia require project. Without project Netia does not enable | |||
that function. | |||
[[Image:Netia-SA_-_SIP_Provider_Compatibility_Test_0.jpg]] | |||
== Current test state == | == Current test state == | ||
{{Template:Compat Status "tested"}} | |||
{{Template:Compat Status "in progress"}} | <!-- {{Template:Compat Status "in progress"}} --> | ||
<!--{{Template:Compat Status "certified"|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}}--> | <!-- {{Template:Compat Status "certified"|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} --> | ||
<!-- {{Template:Compat Status "rejected"}} --> | <!-- {{Template:Compat Status "rejected"}} --> | ||
Testing of this product has been finalized | Testing of this product has been finalized May of 2012. | ||
== Testing Enviroment == | == Testing Enviroment == | ||
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[[Image:HFO_SIP_Compatibility_Test_5.PNG]] | [[Image:HFO_SIP_Compatibility_Test_5.PNG]] | ||
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. This SIP Provider have NAT detection mechanism that allow us not to use STUN or Media-Relay at the SIP interface. | ||
== Test Results == | == Test Results == | ||
Line 50: | Line 72: | ||
|---- | |---- | ||
|'''Call using g711a''' | |'''Call using g711a''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|'''Call using g711u''' | |'''Call using g711u''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|Call using g723 | |Call using g723 | ||
| | |No | ||
|---- | |---- | ||
|Call using g729 | |Call using g729 | ||
| | |Yes | ||
|---- | |||
|Call using g722 | |||
|No | |||
|---- | |---- | ||
|Overlapped sending | |Overlapped sending | ||
| | |No | ||
|---- | |---- | ||
|'''Early media channel''' | |'''Early media channel''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|Fax using T.38 | |Fax using T.38 | ||
| | |No | ||
|---- | |---- | ||
|CGPN can be suppressed | |CGPN can be suppressed | ||
| | |Yes | ||
|---- | |---- | ||
|CLIP no screening | |CLIP no screening | ||
| | |No | ||
|---- | |---- | ||
|Reverse Media Negotiation | |Reverse Media Negotiation | ||
| | |Yes | ||
|---- | |---- | ||
|'''Long time call possible''' | |'''Long time call possible''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|'''External Transfer''' | |'''External Transfer''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|NAT Detection | |NAT Detection | ||
| | |Yes | ||
|---- | |---- | ||
|'''Voice Quality OK?''' | |'''Voice Quality OK?''' | ||
|''' | |'''Yes''' | ||
|---- | |||
|Redundacy Mechanism? | |||
|Yes | |||
|} | |} | ||
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|---- | |---- | ||
|'''Inbound(Provider -> Innovaphone)''' | |'''Inbound(Provider -> Innovaphone)''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|'''Outbound(Innovaphone -> Provider)''' | |'''Outbound(Innovaphone -> Provider)''' | ||
|''' | |'''Yes''' | ||
|} | |} | ||
Line 112: | Line 140: | ||
|---- | |---- | ||
|'''DTMF tones sent correctly''' | |'''DTMF tones sent correctly''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|DTMF tones sent correctly via SIP-Info''' | |DTMF tones sent correctly via SIP-Info''' | ||
| | |No | ||
|---- | |---- | ||
|'''DTMF tones received correctly''' | |'''DTMF tones received correctly''' | ||
|''' | |'''Yes''' | ||
|} | |} | ||
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|---- | |---- | ||
|'''Call can be put on hold''' | |'''Call can be put on hold''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|Held end hears music on hold / announcement from PBX | |Held end hears music on hold / announcement from PBX | ||
| | |Yes | ||
|} | |} | ||
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|---- | |---- | ||
|'''Call can be transferred''' | |'''Call can be transferred''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|Held end hears music on hold | |Held end hears music on hold | ||
| | |Yes | ||
|} | |} | ||
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|---- | |---- | ||
|'''Call can be transferred''' | |'''Call can be transferred''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|Held end hears music on hold or dialling tone | |Held end hears music on hold or dialling tone | ||
| | |Yes | ||
|---- | |---- | ||
|'''Call returns to transferring device if the third Endpoint is not available''' | |'''Call returns to transferring device if the third Endpoint is not available''' | ||
|''' | |'''Yes''' | ||
|} | |} | ||
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|---- | |---- | ||
|Call can be transferred | |Call can be transferred | ||
| | |Yes | ||
|---- | |---- | ||
|Held end hears dialling tone | |Held end hears dialling tone | ||
| | |Yes | ||
|} | |} | ||
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|---- | |---- | ||
|'''Caller can make a call to a Broadcast Group''' | |'''Caller can make a call to a Broadcast Group''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|'''Caller can make a call to a Waiting Queue''' | |'''Caller can make a call to a Waiting Queue''' | ||
|''' | |'''Yes''' | ||
|---- | |---- | ||
|'''Announcement if nobody picks up the call''' | |'''Announcement if nobody picks up the call''' | ||
|''' | |'''Yes''' | ||
|} | |} | ||
== Configuration == | == Configuration == | ||
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[[Image:Netia-SA_-_SIP_Provider_Compatibility_Test_1.png]] | [[Image:Netia-SA_-_SIP_Provider_Compatibility_Test_1.png]] | ||
*Media-Relay is optional | |||
*STUN Server is optional | |||
=== Number Mapping === | === Number Mapping === | ||
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[[Image:Netia-SA_-_SIP_Provider_Compatibility_Test_2.png]] | [[Image:Netia-SA_-_SIP_Provider_Compatibility_Test_2.png]] | ||
*Note "xy" was used to hide the real number used for SIP Test Report. | |||
=== Route Settings === | |||
[[Image:Netia-SA_-_SIP_Provider_Compatibility_Test_3.png]] | |||
*Since the Provider don't support Overlap sending, the flag "force enblock" must be selected. It's recommended the use of "Interworking QSIG/SIP" option in the routes too. | |||
== | == Related Articles == | ||
[[Howto:Poland_-_Netia-S.A._-_SIP_Provider|Poland - Netia SA - SIP Provider]] |
Latest revision as of 09:53, 13 June 2012
Netia S.A. innovaphone Compatibility Test Report
Summary
SIP Provider: Netia.S.A.
The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.
That beeing said, the provider has achieved 88% of all possible test points (110 on total of 125 Points). For more information on the test rating, please refer to Test Description
They are two methods of setting up a SIP trunk with Netia:
- Trunk without Registration - Requires to provide the SIP Provider the Client IP Public address that will make the trunk (SBC or PBX).
- Trunk with Registration - Requires only right credentials (user and password) don't need to know the IP address of client.
We tested the second method. SIP Trunk with Registration.
- Features:
- Direct Dial In
- DTMF
- NAT Detection
- Number presentation (CLIR) could be suppressed
- Redundancy
- Supported Codecs by the provider
- G711 a/u
- G729
Note: T.38 FAX was tested with this provider during the tests. This feature didn't worked as expected so is listed as "Not Supported" however in future fixes could work with Innovaphone probably.
- Redundancy Methods:
A) Using Session Agent Group on SBC - only static SIP Trunk (SIP Trunk without registration).
B) Using Enterprise Group on AS - using OTG/DTG Identity.
Note: For redundancy Netia require project. Without project Netia does not enable that function.
Current test state
The tests for this product have been completed.
Testing of this product has been finalized May of 2012.
Testing Enviroment
Scenario NAT
This scenario describes a setup where the PBX and phones are in a private network. This SIP Provider have NAT detection mechanism that allow us not to use STUN or Media-Relay at the SIP interface.
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.
Tested feature | Result |
---|---|
Call using g711a | Yes |
Call using g711u | Yes |
Call using g723 | No |
Call using g729 | Yes |
Call using g722 | No |
Overlapped sending | No |
Early media channel | Yes |
Fax using T.38 | No |
CGPN can be suppressed | Yes |
CLIP no screening | No |
Reverse Media Negotiation | Yes |
Long time call possible | Yes |
External Transfer | Yes |
NAT Detection | Yes |
Voice Quality OK? | Yes |
Redundacy Mechanism? | Yes |
Direct Dial In
Tested feature | Result |
---|---|
Inbound(Provider -> Innovaphone) | Yes |
Outbound(Innovaphone -> Provider) | Yes |
DTMF
Tested feature | Result |
---|---|
DTMF tones sent correctly | Yes |
DTMF tones sent correctly via SIP-Info | No |
DTMF tones received correctly | Yes |
Hold/Retrieve
Tested feature | Result |
---|---|
Call can be put on hold | Yes |
Held end hears music on hold / announcement from PBX | Yes |
Transfer with consultation
Tested feature | Result |
---|---|
Call can be transferred | Yes |
Held end hears music on hold | Yes |
Transfer with consultation (alerting only)
Tested feature | Result |
---|---|
Call can be transferred | Yes |
Held end hears music on hold or dialling tone | Yes |
Call returns to transferring device if the third Endpoint is not available | Yes |
Blind Transfer
Tested feature | Result |
---|---|
Call can be transferred | Yes |
Held end hears dialling tone | Yes |
Broadcast Group & Waiting Queue
Tested feature | Result |
---|---|
Caller can make a call to a Broadcast Group | Yes |
Caller can make a call to a Waiting Queue | Yes |
Announcement if nobody picks up the call | Yes |
Configuration
Firmware version
All innovaphone devices use V9hf11 as firmware.
SIP - Trunk
- Media-Relay is optional
- STUN Server is optional
Number Mapping
- Note "xy" was used to hide the real number used for SIP Test Report.
Route Settings
- Since the Provider don't support Overlap sending, the flag "force enblock" must be selected. It's recommended the use of "Interworking QSIG/SIP" option in the routes too.