Howto:Skype Connect - SIP Testreport: Difference between revisions

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'''SIP Provider: Skype'''
'''SIP Provider: Skype'''


<!--
* The mandatory feature '''Early Media''' is not supported by Skype Connect, the provider never sends any Session Progress message to the PBX before call establishment.
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]].  
* Skype connect rejects calls with Diverting-info, special configuration is required<ref name="div">It is necessary to enable the ''Set Calling=Diverting No'' option on the ''Trunk Line'' object. If not, a diverted call is rejected by the provider with ''SIP 403 Forbidden'' because it contains ''Diverting Info''.</ref>.




...
*Caller identification using your Skype Number is available in:


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#Summary|Test Description]]
**USA
**UK
**Chile
**Denmark
**Estonia
**Hong Kong
**Poland
**Sweden


* Features:
*As alternative we could verify a landline number using the ''Verify Procedure'' of Skype or use a mobile phone as Caller ID. However even these verified numbers are only available for limited countries.


** Direct Dial In
** Fax over IP (T.38)
** DTMF
* Supported Codecs by the provider
** G711
** G729
** G723
** G726
** T.38 UDP
-->
== Current test state ==
== Current test state ==


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<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


Testing of this product has been finalized January 20th, 2012.
Testing of this product has been finalized December 20th, 2012.
 
 
Mandatory feature like Early Media it's not supported by Skype Connect, Provider never sends any Session Progress message to the PBX before call establishment, this will not allow users to receive audio messages from Providers when calling out.
 
Other found issue was that Skype connect rejects calls with Diverting-info, to overcome this problem we have to enable the option "Set Calling=Diverting No" on Trunkline Object.
 
Caller identification using your Skype Number is available in:
 
USA
UK
Chile
Denmark
Estonia
Hong Kong
Poland
Sweden
 
As alternative we could verify a landline number using the Verify Procedure of Skype that could take up 3 days or use Mobile Phone as Caller ID but still only available for limited countries.


== Testing Enviroment ==
== Testing Enviroment ==
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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.  


* 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.


== Test Results ==
== Test Results ==
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|----
|----
|'''early media channel'''
|'''early media channel'''
|'''NOK''',No SDP in 18x messages.
|'''NOK''', no SDP in 18x messages.
|----
|----
|Fax using T.38
|Fax using T.38
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|----
|----
|'''Call can be forward'''
|'''Call can be forward'''
|'''OK*'''
|'''OK''' <ref name='div'/>
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
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|}
|}


"*" - It's necessary to set up on Trunline Object "Set Calling=Diverting No".If not call is rejected by Provider with SIP 403 Forbidden due Diverting Info.
 


=== CFNR/Blind Transfer (alerting only)===
=== CFNR/Blind Transfer (alerting only)===
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|----
|----
|'''Call can be transferred or forward'''
|'''Call can be transferred or forward'''
|'''OK*'''
|'''OK''' <ref name='div'/>
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
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|}
|}


"*" - It's necessary to set up on Trunline Object "Set Calling=Diverting No".If not call is rejected by Provider with SIP 403 Forbidden due Diverting Info.
 


The following tests are made to test if call transfer is working.
The following tests are made to test if call transfer is working.
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[[Image:Skype_Connect_-_SIP_Testreport_2.png]]
[[Image:Skype_Connect_-_SIP_Testreport_2.png]]


* Skype Caller ID for outgoing calls it's defined on the Skype Manager page that's why we don't define any CGPN Out value.
The Skype ''Caller ID'' for outgoing calls is defined on the ''Skype Manager'' page. For this reason, we don't define any ''CGPN Out'' value.


=== Route Settings ===
=== Route Settings ===
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[[Image:Skype_Connect_-_SIP_Testreport_3.png]]
[[Image:Skype_Connect_-_SIP_Testreport_3.png]]


* ForceEnblock it's necessary for outgoing calls.
You must set the ''Force enblock'' (b) option on the route for outgoing calls.


=== Known Issues ===
== Known Issues ==


[[Image:Skype_Connect_-_SIP_Testreport_4.png]]
[[Image:Skype_Connect_-_SIP_Testreport_4.png]]<ref name="div"/>


* It's necessary to set up on Trunline Object "Set Calling=Diverting No".If not call is rejected by Provider with SIP 403 Forbidden due Diverting Info. Diverting info it's always sent even if we don't use Interwork QSIG/SIG option.
<references/>


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

Latest revision as of 09:23, 13 February 2018

Innovaphone Compatibility Test Report

Summary

SIP Provider: Skype

  • The mandatory feature Early Media is not supported by Skype Connect, the provider never sends any Session Progress message to the PBX before call establishment.
  • Skype connect rejects calls with Diverting-info, special configuration is required[1].


  • Caller identification using your Skype Number is available in:
    • USA
    • UK
    • Chile
    • Denmark
    • Estonia
    • Hong Kong
    • Poland
    • Sweden
  • As alternative we could verify a landline number using the Verify Procedure of Skype or use a mobile phone as Caller ID. However even these verified numbers are only available for limited countries.

Current test state

The tests for this product could not be completed or not all mandatory tests were passed. See the Summary section for more details.

Testing of this product has been finalized December 20th, 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 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 NOK
call using g729 OK
call using g722 NOK
Overlapped sending NOK
early media channel NOK, no SDP in 18x messages.
Fax using T.38 Not supported as written on product description, not tested
Reverse Media Negotiation OK
CGPN can be suppressed OK
CLIP no screening NOK
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection OK
Redundancy Not tested
SIP over TCP NOK, no _tcp serv ice record in skype.com domain
Voice Quality OK? OK

Direct Dial In

Tested feature Result
Inbound(Provider -> Innovaphone via SkypeIn Number) OK
Inbound(Provider -> Innovaphone via Skype User) 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

CFU/ CFB Transfer

Tested feature Result
Call can be forward OK [1]
Held end hears dialling tone OK


CFNR/Blind Transfer (alerting only)

Tested feature Result
Call can be transferred or forward OK [1]
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 hf19 as firmware.

SIP - Trunk

Number Mapping

The Skype Caller ID for outgoing calls is defined on the Skype Manager page. For this reason, we don't define any CGPN Out value.

Route Settings

You must set the Force enblock (b) option on the route for outgoing calls.

Known Issues

[1]

  1. ↑ 1.0 1.1 1.2 1.3 It is necessary to enable the Set Calling=Diverting No option on the Trunk Line object. If not, a diverted call is rejected by the provider with SIP 403 Forbidden because it contains Diverting Info.
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