Howto:QSC VoIP connect - SIP Provider Compatibility Test: Difference between revisions

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(New page: '''Innovaphone Compatibility Test Report''' == Summary == '''SIP Provider: QSC''' == Current test state == <!-- {{Template:Compat Status "tested"(sip provider)}} --> {{Template:Compat...)
 
 
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'''SIP Provider: QSC'''
'''SIP Provider: QSC'''


* Features:
** Direct Dial In
** DTMF
* Supported Codecs by the provider
** G711
** G729
** G723
** T.38
Technically, QSC ''VoIP Connect'' meets all requirements for a recommended SIP trunk.  However,
''VoIP connect'' is a wholesale SIP trunk product. It implements phone calls for many customers (i.e. many subscriber numbers) on a single SIP trunk (a.k.a. ''fat pipe'').  The operator of such a trunk has to meet the legal obligations of a telecommunication carrier, as imposed by the laws in the respective country. As a result, users of this product need to be prepared to fulfil all such requirements.  If you plan to use the ''VoIP connect'' trunk, we recommend contacting innovaphone [mailto:presales@innovaphone.com pre-sales] for discussing your concept/planned installation.
For customers who do not want to take over carrier responsibilities, we recommend using QSC's [[Howto:QSC_IPfonie_extended_-_SIP_Provider_Compatibility_Test | IPfonie extended]] product instead.


== Current test state ==
== Current test state ==
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
{{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 November 30th, 2007. -->
Testing of this product has been finalized November 19th, 2009.


== Testing Enviroment ==
== Testing Enviroment ==


[[Image:QSC VoIP connect - SIP Provider Compatibility Test 1.png]]
In this scenario the PBX is in the public network. The registered phones are alternatively in private or public networks. STUN is not used, the SIP provider is responsible for handling SIP messages with private IP addresses. Moreover mediarelay is not configured, resulting in RTP streams not passing the PBX but going directly between the SIP provider and the used phone.


== Test Results ==
== Test Results ==
Line 29: Line 47:
|----
|----
|'''call using g711a'''
|'''call using g711a'''
|'''?'''
|'''Yes'''
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|'''?'''
|'''Yes'''
|----
|----
|call using g723
|call using g723
|?
|Yes
|----
|----
|call using g729
|call using g729
|?
|Yes
|----
|----
|Overlapped sending
|Overlapped sending
|?
|No
|----
|----
|'''early media channel'''
|'''early media channel'''
|'''?'''
|'''Yes'''
|----
|----
|Fax using T.38
|Fax using T.38
|?
|Yes
|----
|CGPN can be suppressed
|Yes (with special flag)
|----
|----
|CGPN can be supressed
|'''CLIP no screening'''
|?
|'''No'''
|----
|----
|'''Reverse Media Negotiaton'''
|'''Reverse Media Negotiaton'''
|'''?'''
|'''Yes'''
|----
|Long time call possible(>30 min)
|'''Yes'''
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|'''?'''
|'''Yes'''
|}
|}


Line 66: Line 90:
|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone)'''
|'''?'''
|'''Yes'''
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|'''?'''
|'''Yes'''
|}
|}


Line 79: Line 103:
|----
|----
|'''DTMF tones sent correctly'''
|'''DTMF tones sent correctly'''
|'''?'''
|'''Yes'''
|----
|----
|'''DTMF tones received correctly'''
|'''DTMF tones received correctly'''
|'''?'''
|'''Yes'''
|}
|}


Line 92: Line 116:
|----
|----
|'''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
|----
|Held end hears music on hold / announcement from provider
|?
|}
|}


Line 108: Line 129:
|----
|----
|'''Call can be transfered'''
|'''Call can be transfered'''
|'''?'''
|'''Yes'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|?
|Yes
|----
|----
|'''Call returns to transferring device if the third'''  
|'''Call returns to transferring device if the third'''  
'''Endpoint is not available'''
'''Endpoint is not available'''
|'''?'''
|'''Yes'''
|}
|}


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|----
|----
|'''Call can be transfered'''
|'''Call can be transfered'''
|'''?'''
|'''Yes'''
|----
|----
|Held end hears music on hold or dialing tone
|Held end hears music on hold or dialing tone
|?
|Yes
|----
|----
|'''Call returns to transferring device if the third'''  
|'''Call returns to transferring device if the third'''  
'''Endpoint is not available'''
'''Endpoint is not available'''
|'''?'''
|'''Yes'''
|}
|}


Line 142: Line 163:
|----
|----
|Call can be transfered
|Call can be transfered
|?
|Yes
|----
|----
|Held end hears dialing tone
|Held end hears dialing tone
|?
|Yes
|}
|}


Line 155: Line 176:
|----
|----
|'''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 ==


===Firmware version===
All innovaphone devices use V8 firmware.
=== SIP - Trunk ===
First of all the SIP Trunk must be configured. Since QSC authenticates a user account only by IP - Address, you cannot use the normal SIP-Gateway object. You need to configure a GW without registration and route the calls to the PBX.
[[Image:QSC VoIP connect - SIP Provider Compatibility Test 2.PNG]]
=== Number Mapping ===
[[Image:QSC VoIP connect - SIP Provider Compatibility Test 3.PNG]]
=== Route Settings ===
Because QSC, as most SIP - Providers too, doesn't support overlap sending, you must enable the block-wise sending of the phone number. You can do this by enabling ''Force enblock'' in your routes.
The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this check-box is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.
===CLIR===
To suppress the CGPN for outbound calls a config line option must be activated. Please enter the following lines in your browser:
http://PBX-IP-address/!config add SIP /pai
http://PBX-IP-address/!config write
http://PBX-IP-address/!config activate


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

Latest revision as of 17:53, 3 March 2010

Innovaphone Compatibility Test Report

Summary

SIP Provider: QSC

  • Features:
    • Direct Dial In
    • DTMF
  • Supported Codecs by the provider
    • G711
    • G729
    • G723
    • T.38

Technically, QSC VoIP Connect meets all requirements for a recommended SIP trunk. However, VoIP connect is a wholesale SIP trunk product. It implements phone calls for many customers (i.e. many subscriber numbers) on a single SIP trunk (a.k.a. fat pipe). The operator of such a trunk has to meet the legal obligations of a telecommunication carrier, as imposed by the laws in the respective country. As a result, users of this product need to be prepared to fulfil all such requirements. If you plan to use the VoIP connect trunk, we recommend contacting innovaphone pre-sales for discussing your concept/planned installation.

For customers who do not want to take over carrier responsibilities, we recommend using QSC's IPfonie extended product instead.

Current test state

The tests for this product have been completed.

Testing of this product has been finalized November 19th, 2009.

Testing Enviroment

QSC VoIP connect - SIP Provider Compatibility Test 1.png

In this scenario the PBX is in the public network. The registered phones are alternatively in private or public networks. STUN is not used, the SIP provider is responsible for handling SIP messages with private IP addresses. Moreover mediarelay is not configured, resulting in RTP streams not passing the PBX but going directly between the SIP provider and the used phone.

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 Yes
call using g711u Yes
call using g723 Yes
call using g729 Yes
Overlapped sending No
early media channel Yes
Fax using T.38 Yes
CGPN can be suppressed Yes (with special flag)
CLIP no screening No
Reverse Media Negotiaton Yes
Long time call possible(>30 min) Yes
Voice Quality OK? 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 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 transfered Yes
Held end hears music on hold Yes
Call returns to transferring device if the third

Endpoint is not available

Yes

Transfer with consultation (alerting only)

Tested feature Result
Call can be transfered Yes
Held end hears music on hold or dialing tone Yes
Call returns to transferring device if the third

Endpoint is not available

Yes

Blind Transfer

Tested feature Result
Call can be transfered Yes
Held end hears dialing 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 V8 firmware.

SIP - Trunk

First of all the SIP Trunk must be configured. Since QSC authenticates a user account only by IP - Address, you cannot use the normal SIP-Gateway object. You need to configure a GW without registration and route the calls to the PBX.

QSC VoIP connect - SIP Provider Compatibility Test 2.PNG

Number Mapping

QSC VoIP connect - SIP Provider Compatibility Test 3.PNG

Route Settings

Because QSC, as most SIP - Providers too, doesn't support overlap sending, you must enable the block-wise sending of the phone number. You can do this by enabling Force enblock in your routes.

The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this check-box is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.

CLIR

To suppress the CGPN for outbound calls a config line option must be activated. Please enter the following lines in your browser: 

http://PBX-IP-address/!config add SIP /pai
http://PBX-IP-address/!config write
http://PBX-IP-address/!config activate
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