Howto:QSC IPfonie extended - SIP Provider Compatibility Test: Difference between revisions
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'''SIP Provider: QSC''' | '''SIP Provider: QSC''' | ||
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]]. | |||
Please note that in order to use [[Howto:How_does_"CLIP_no_screening"_work%3F | Clip no screening]], some [[Howto:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test#Known_Issues|extra configuration]] must be done. | |||
That being said, the provider has achieved 96.2% of all possible test points (128 on 133). For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]] | |||
* Features: | * Features: | ||
** T.38 IP Fax | |||
** Direct Dial In | ** Direct Dial In | ||
** DTMF | ** DTMF | ||
** CGPN can be suppressed | |||
** CLIP No Screening | |||
** SIP over TCP | |||
* Supported Codecs by the provider | * Supported Codecs by the provider | ||
** | ** G711a/u | ||
** G722 | |||
** G729 | ** G729 | ||
** G723 | ** G723 | ||
** T.38 | ** T.38 UDP | ||
== Current test state == | |||
<!--{{Template:Compat Status "planned"}} --> | |||
<!--{{Template:Compat Status " | |||
<!-- {{Template:Compat Status "in progress"}} --> | <!-- {{Template:Compat Status "in progress"}} --> | ||
{{Template: | <!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} --> | ||
<!-- {{Template:Compat Status "tested"}}--> | <!-- {{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 "tested"(sip provider)}} --> | |||
<!-- {{Template:Compat Status "rejected"}} --> | <!-- {{Template:Compat Status "rejected"}} --> | ||
<!-- {{Template:Compat Status "customer-testimonial"}} --> | |||
{{Template:Compat Status "certified"|certificate=QSC_-_IPfonie_extended_-_SIP_Provider_-_product-cert.pdf}} | |||
Testing of this product has been finalized | Testing of this product has been finalized October 11th, 2012. | ||
== Testing Enviroment == | == Testing Enviroment == | ||
[[Image: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, no STUN and without exclusive coder. | |||
== 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. | ||
=== Basic Call === | === Basic Call === | ||
Line 52: | 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 | |||
|OK* | |||
|---- | |---- | ||
|Overlapped sending | |Overlapped sending | ||
| | |NOK | ||
|---- | |---- | ||
|'''early media channel''' | |'''early media channel''' | ||
|''' | |'''OK''' | ||
|---- | |---- | ||
|Fax using T.38 | |Fax using T.38 | ||
| | |OK | ||
|---- | |||
|Reverse Media Negotiation | |||
|OK | |||
|---- | |---- | ||
|CGPN can be | |CGPN can be suppressed | ||
| | |OK | ||
|---- | |---- | ||
|CLIP no screening | |CLIP no screening | ||
| | |OK | ||
|---- | |---- | ||
|''' | |'''Long time call possible(>30 min)''' | ||
|''' | |'''OK''' | ||
|---- | |---- | ||
|'''External Transfer''' | |'''External Transfer''' | ||
|''' | |'''OK''' | ||
|---- | |||
|NAT Detection | |||
|OK | |||
|---- | |||
|Redundancy | |||
|OK | |||
|---- | |||
|SIP over TCP | |||
|OK | |||
|---- | |---- | ||
|'''Voice Quality OK?''' | |'''Voice Quality OK?''' | ||
|''' | |'''OK''' | ||
|} | |} | ||
"*" - G722 is only supported if the remote endpoint also supports this codec (QSC doesn't do transcoding of the calls). | |||
=== Direct Dial In === | === Direct Dial In === | ||
Line 95: | Line 120: | ||
|---- | |---- | ||
|'''Inbound(Provider -> Innovaphone)''' | |'''Inbound(Provider -> Innovaphone)''' | ||
|''' | |'''OK''' | ||
|---- | |---- | ||
|'''Outbound(Innovaphone -> Provider)''' | |'''Outbound(Innovaphone -> Provider)''' | ||
|''' | |'''OK''' | ||
|} | |} | ||
Line 108: | Line 133: | ||
|---- | |---- | ||
|'''DTMF tones sent correctly''' | |'''DTMF tones sent correctly''' | ||
|''' | |'''OK''' | ||
|---- | |||
|DTMF tones sent correctly via SIP-Info | |||
|NOK | |||
|---- | |---- | ||
|'''DTMF tones received correctly''' | |'''DTMF tones received correctly''' | ||
|''' | |'''OK''' | ||
|} | |} | ||
Line 121: | Line 149: | ||
|---- | |---- | ||
|'''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 136: | Line 161: | ||
!Result | !Result | ||
|---- | |---- | ||
|'''Call can be | |'''Call can be transferred''' | ||
|''' | |'''OK''' | ||
|---- | |---- | ||
|Held end hears music on hold | |Held end hears music on hold | ||
| | |OK | ||
|} | |||
The following tests are made to test if call transfer is working. | |||
{| border="1" | |||
!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 | |||
|} | |} | ||
Line 153: | Line 202: | ||
!Result | !Result | ||
|---- | |---- | ||
|'''Call can be | |'''Call can be transferred''' | ||
|''' | |'''OK''' | ||
|---- | |---- | ||
|Held end hears music on hold or | |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''' | ||
|} | |||
The following tests are made to test if call transfer is working. | |||
{| border="1" | |||
!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 | |||
|} | |} | ||
Line 170: | Line 247: | ||
!Result | !Result | ||
|---- | |---- | ||
|Call can be | |Call can be transferred | ||
| | |OK | ||
|---- | |||
|Held end hears dialling tone | |||
|OK | |||
|} | |||
The following tests are made to test if call transfer is working. | |||
{| border="1" | |||
!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 | |||
|} | |||
=== Blind Transfer (alerting only)=== | |||
{| border="1" | |||
!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. | |||
{| border="1" | |||
!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 | |||
|} | |} | ||
Line 184: | Line 312: | ||
|---- | |---- | ||
|'''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''' | ||
|} | |} | ||
== Configuration == | == Configuration == | ||
=== Firmware version === | ===Firmware version=== | ||
All innovaphone devices use V9 hotfix 16 build 9.061101 as firmware. | |||
=== SIP - Trunk === | === SIP - Trunk === | ||
Here's the configuration of the SIP gateway interface. | |||
[[Image:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test_1.png]] | |||
[[Image: | |||
=== Number Mapping === | === Number Mapping === | ||
[[Image:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test_2.png]] | |||
[[Image: | |||
=== Route Settings === | === Route Settings === | ||
[[Image:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test_3.png]] | |||
[[Image: | |||
Force Enblock Setting it's required for outgoing calls. QSIG/SIP Interworking setting it's recommended. | |||
=== | === Redundancy === | ||
* QSC allows more than one registration on the same account. Incoming calls will be delivered on the first device that registered on the account. If the first device goes down or looses its registration, then incoming calls are delivered on the 2nd registered device. When the initial device comes back online then calls will delivered again to it. | |||
=== Known Issues === | |||
* In order to CLIP No Screening work properly we must set option to send SIP Address P-Asserted Identity like below. | |||
http://x.x.x.x/!config add SIP /pai | |||
http://x.x.x.x/!config write | |||
http://x.x.x.x/!config activate | |||
[[Category:Compat|{{PAGENAME}}]] | [[Category:Compat|{{PAGENAME}}]] |
Latest revision as of 14:52, 18 June 2014
Innovaphone Compatibility Test Report
Summary
SIP Provider: QSC
The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.
Please note that in order to use Clip no screening, some extra configuration must be done.
That being said, the provider has achieved 96.2% of all possible test points (128 on 133). For more information on the test rating, please refer to Test Description
- Features:
- T.38 IP Fax
- Direct Dial In
- DTMF
- CGPN can be suppressed
- CLIP No Screening
- SIP over TCP
- Supported Codecs by the provider
- G711a/u
- G722
- G729
- G723
- T.38 UDP
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 October 11th, 2012.
Testing Enviroment
This scenario describes a setup where the PBX and phones are in a private network.
The SIP trunk is configured without Media Relay, no STUN and without exclusive coder.
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 | OK* |
Overlapped sending | NOK |
early media channel | OK |
Fax using T.38 | OK |
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 | OK |
SIP over TCP | OK |
Voice Quality OK? | OK |
"*" - G722 is only supported if the remote endpoint also supports this codec (QSC doesn't do transcoding of the calls).
Direct Dial In
Tested feature | Result |
---|---|
Inbound(Provider -> Innovaphone) | OK |
Outbound(Innovaphone -> Provider) | 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 |
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 |
Blind Transfer (alerting only)
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 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 hotfix 16 build 9.061101 as firmware.
SIP - Trunk
Here's the configuration of the SIP gateway interface.
Number Mapping
Route Settings
Force Enblock Setting it's required for outgoing calls. QSIG/SIP Interworking setting it's recommended.
Redundancy
- QSC allows more than one registration on the same account. Incoming calls will be delivered on the first device that registered on the account. If the first device goes down or looses its registration, then incoming calls are delivered on the 2nd registered device. When the initial device comes back online then calls will delivered again to it.
Known Issues
- In order to CLIP No Screening work properly we must set option to send SIP Address P-Asserted Identity like below.
http://x.x.x.x/!config add SIP /pai http://x.x.x.x/!config write http://x.x.x.x/!config activate