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]].  


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 the Trunkline object.


...
*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
* Features:
**Chile
**Denmark
**Estonia
**Hong Kong
**Poland
**Sweden


** Direct Dial In
*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.
** 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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Testing of this product has been finalized December 20th, 2012.
Testing of this product has been finalized December 20th, 2012.
== Summary ==
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
Line 308: Line 284:
|}
|}


"*" - 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.
"*" - It's necessary to enable on the Trunkline object the "Set Calling=Diverting No" option. If not the call is rejected by the provider with ''SIP 403 Forbidden'' because of it contains ''Diverting Info''.


=== CFNR/Blind Transfer (alerting only)===
=== CFNR/Blind Transfer (alerting only)===
Line 323: Line 299:
|}
|}


"*" - 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.
"*" - It's necessary to enable on the Trunkline object the "Set Calling=Diverting No" option. If not the call is rejected by the provider with ''SIP 403 Forbidden'' because of it contains ''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.
Line 382: Line 358:


[[Image:Skype_Connect_-_SIP_Testreport_4.png]]
[[Image:Skype_Connect_-_SIP_Testreport_4.png]]
 
"*" - It's necessary to enable on the Trunkline object the "Set Calling=Diverting No" option. If not the call is rejected by the provider with ''SIP 403 Forbidden'' because of it contains ''Diverting Info''. Diverting info it's always sent even if we don't use the ''Interwork QSIG/SIG'' option.
* 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.


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

Revision as of 17:09, 15 February 2013

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.

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 the 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 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

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 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*
Held end hears dialling tone OK

"*" - It's necessary to enable on the Trunkline object the "Set Calling=Diverting No" option. If not the call is rejected by the provider with SIP 403 Forbidden because of it contains Diverting Info.

CFNR/Blind Transfer (alerting only)

Tested feature Result
Call can be transferred or forward OK*
Held end hears dialling tone OK

"*" - It's necessary to enable on the Trunkline object the "Set Calling=Diverting No" option. If not the call is rejected by the provider with SIP 403 Forbidden because of it contains Diverting Info.

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

Skype Connect - SIP Testreport 1.png

Number Mapping

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.

Route Settings

Skype Connect - SIP Testreport 3.png

  • ForceEnblock it's necessary for outgoing calls.

Known Issues

Skype Connect - SIP Testreport 4.png "*" - It's necessary to enable on the Trunkline object the "Set Calling=Diverting No" option. If not the call is rejected by the provider with SIP 403 Forbidden because of it contains Diverting Info. Diverting info it's always sent even if we don't use the Interwork QSIG/SIG option.

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