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 ==


<!--{{Template:Compat Status "planned"}} -->
<!--{{Template:Compat Status "planned"}} -->
{{Template:Compat Status "in progress"}}
<!--{{Template:Compat Status "in progress"}}-->
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"engineered_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{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_"rec._prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "rejected"}} -->
{{Template:Compat Status "rejected"}}
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


<!-- Testing of this product has been finalized January 1st, 1970. -->
Testing of this product has been finalized December 20th, 2012.


== Testing Enviroment ==
== Testing Enviroment ==
Line 46: Line 41:
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.  


There are 3 major configuration variants of the SIP trunk, which one is used depends on the test results. The variants are:
* the SIP trunk is configured without Media Relay and without exclusive coder.
 
* 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 with Media Relay and STUN but without exclusive coder. This is the case when the test for "NAT Traversal" fails
* the SIP trunk is configured with Media Relay and exclusive coder. This is the case when the test for "Reverse Media Negotiation" fails
 
The test scenario should describe which SIP trunk configuration is needed.


== 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.
{{FIXME|reason=Note: Possible test results are Ok, Nok, Not tested, N/A(not available). Ok, Nok, N/A are self explanatory. "Not tested" is more a temporary note - if its still in the test after finishing, then the summary should explain why the feature was not tested. Remove this note in the final report}}


=== Basic Call ===
=== Basic Call ===
Line 67: Line 54:
|----
|----
|'''call using g711a'''
|'''call using g711a'''
|'''Nok, if call rejected at PSTN - no Cancel from PSTN side.''' In addition calls to the PSTN were signalled also to completely unknown PSTN number, were the call was answered and played to calling phone while PSTN phone was still ringing.
|'''OK'''
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|Nok - outbound call one way media
|'''OK'''
|----
|----
|call using g723
|call using g723
|Nok
|NOK
|----
|----
|call using g729
|call using g729
|Ok
|OK
|----
|----
|call using g722
|call using g722
|Nok
|NOK
|----
|----
|Overlapped sending
|Overlapped sending
|Nok
|NOK
|----
|----
|'''early media channel'''
|'''early media channel'''
|Could not be tested, calls to unknown numbers are not signalled correctly at calling phone. No SDP in 18x messages, therefore NOK
|'''NOK''', no SDP in 18x messages.
|----
|----
|Fax using T.38
|Fax using T.38
Line 91: Line 78:
|----
|----
|Reverse Media Negotiation
|Reverse Media Negotiation
|Ok
|OK
|----
|----
|CGPN can be suppressed
|CGPN can be suppressed
|CGPN not displayed
|OK
|----
|----
|CLIP no screening
|CLIP no screening
|CGPN not displayed
|NOK
|----
|----
|'''Long time call possible(>30 min)'''
|'''Long time call possible(>30 min)'''
|Not tested
|'''OK'''
|----
|----
|'''External Transfer'''
|'''External Transfer'''
|Not tested
|'''OK'''
|----
|----
|NAT Detection
|NAT Detection
|Ok
|OK
|----
|----
|Redundancy
|Redundancy
Line 112: Line 99:
|----
|----
|SIP over TCP
|SIP over TCP
|Nok, no _tcp serv ice record in skype.com domain
|NOK, no _tcp serv ice record in skype.com domain
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
Line 124: Line 111:
!Result  
!Result  
|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone via SkypeIn Number)'''
|Not tested
|'''OK'''
|----
|'''Inbound(Provider -> Innovaphone via Skype User)'''
|'''OK'''
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|Nok, CGPN shown as anonym
|'''OK'''
|}
|}


Line 138: Line 128:
|----
|----
|'''DTMF tones sent correctly'''
|'''DTMF tones sent correctly'''
|Nok via RFC-2833, only inband in RTP-Stream
|'''OK'''
|----
|----
|DTMF tones sent correctly via SIP-Info
|DTMF tones sent correctly via SIP-Info
|Nok
|NOK
|----
|----
|'''DTMF tones received correctly'''
|'''DTMF tones received correctly'''
|Not tested
|'''OK'''
|}
|}


Line 154: Line 144:
|----
|----
|'''Call can be put on hold'''  
|'''Call can be put on hold'''  
|'''Ok'''  
|'''OK'''  
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|Ok
|OK
|}
|}


Line 167: Line 157:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|Ok, only with mediarelay. Without mediarelay no audio after transfer.
|'''OK'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|Ok
|OK
|}
|}


Line 181: Line 171:
|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|}
|}


Line 208: Line 198:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''OK'''
|----
|----
|Held end hears music on hold or dialling tone
|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'''
|}
|}


Line 226: Line 216:
|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|OK
|
|OK
|}
|}


Line 253: Line 243:
|----
|----
|Call can be transferred
|Call can be transferred
|
|OK
|----
|----
|Held end hears dialling tone
|Held end hears dialling tone
|
|OK
|}
|}


Line 266: Line 256:
|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|----
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|----
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|OK
|}
|}


=== Blind Transfer (alerting only)===
=== CFU/ CFB Transfer ===


{| border="1"  
{| border="1"  
Line 287: Line 277:
!Result  
!Result  
|----
|----
|Call can be transferred
|'''Call can be forward'''
|
|'''OK''' <ref name='div'/>
|----
|'''Held end hears dialling tone'''
|'''OK'''
|}
 
 
 
=== CFNR/Blind Transfer (alerting only)===
 
{| border="1"
!Tested feature
!Result
|----
|'''Call can be transferred or forward'''
|'''OK''' <ref name='div'/>
|----
|----
|Held end hears dialling tone
|'''Held end hears dialling tone'''
|
|'''OK'''
|}
|}


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 301: Line 308:
|----
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|OK
|----
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|OK
|----
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|OK
|}
|}


Line 317: Line 324:
|----
|----
|'''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'''
|}
|}


Line 330: Line 337:
===Firmware version===
===Firmware version===


All innovaphone devices use Vx build xx-xxxxx as firmware.
All innovaphone devices use V9 hf19 as firmware.


=== SIP - Trunk ===
=== SIP - Trunk ===


[[Image:Skype_Connect_-_SIP_Testreport_1.png]]


=== Number Mapping ===
=== Number Mapping ===


[[Image:Skype_Connect_-_SIP_Testreport_2.png]]
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 ===


[[Image:Skype_Connect_-_SIP_Testreport_3.png]]
You must set the ''Force enblock'' (b) option on the route for outgoing calls.


=== Media Relay ===
== Known Issues ==


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


=== Fax ===
<references/>


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

Latest revision as of 10:23, 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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