Howto:Skype Connect - SIP Testreport: Difference between revisions

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


The provider '''does not''' support all required innovaphone features and is '''not''' qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|recommended SIP Provider]].  
<!--
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]].  


Incoming calls were not possible at all, therefore the tests were aborted. Moreover the provider seems to have problems with handling multiple registrations on one account.


...
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]]
* Features:
** 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 "certified"|certificate=Product_-_Vendor_-_3rd_Party_Product_-_Desc-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_"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 "customer-testimonial"}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


Testing of this product has been finalized April 20th, 2010.
<!-- Testing of this product has been finalized January 1st, 1970. -->


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


=== Scenario NAT ===
[[Image:HFO_SIP_Compatibility_Test_5.PNG]]
 
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:


[[Image:HFO_SIP_Compatibility_Test_5.PNG]]
* 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


This scenario describes a setup where the PBX and phones are in a private network. The IP800 must use a stun server, in order to send correct SIP - messages. The IP800 works as media relay, all RTP - streams go through the PBX.
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 39: Line 67:
|----
|----
|'''call using g711a'''
|'''call using g711a'''
|'''No'''
|'''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.
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|'''No'''
|Nok - outbound call one way media
|----
|----
|call using g723
|call using g723
|not tested
|Nok
|----
|----
|call using g729
|call using g729
|not tested
|Ok
|----
|call using g722
|Nok
|----
|----
|Overlapped sending
|Overlapped sending
|not tested
|Nok
|----
|----
|'''early media channel'''
|'''early media channel'''
|not tested
|Could not be tested, calls to unknown numbers are not signalled correctly at calling phone. No SDP in 18x messages, therefore NOK
|----
|----
|Fax using T.38
|Fax using T.38
|not tested
|Not supported as written on product description, not tested
|----
|----
|Reverse Media Negotiation
|Reverse Media Negotiation
|not tested
|Ok
|----
|----
|CGPN can be suppressed
|CGPN can be suppressed
|not tested
|CGPN not displayed
|----
|----
|CLIP no screening
|CLIP no screening
|not tested
|CGPN not displayed
|----
|----
|'''Long time call possible(>30 min)'''
|'''Long time call possible(>30 min)'''
|not tested
|Not tested
|----
|----
|'''External Transfer'''
|'''External Transfer'''
|not tested
|Not tested
|----
|----
|NAT Detection
|NAT Detection
|not tested
|Ok
|----
|Redundancy
|Not tested
|----
|SIP over TCP
|Nok, no _tcp serv ice record in skype.com domain
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|'''No'''
|'''OK'''
|}
 
=== Direct Dial In ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|Not tested
|----
|'''Outbound(Innovaphone -> Provider)'''
|Nok, CGPN shown as anonym
|}
 
=== DTMF ===
 
{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|Nok via RFC-2833, only inband in RTP-Stream
|----
|DTMF tones sent correctly via SIP-Info
|Nok
|----
|'''DTMF tones received correctly'''
|Not tested
|}
 
=== Hold/Retrieve ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Ok'''
|----
|Held end hears music on hold / announcement from PBX
|Ok
|}
 
=== Transfer with consultation ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Call can be transferred'''
|Ok, only with mediarelay. Without mediarelay no audio after transfer.
|----
|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.
|
|
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|
|}
 
=== Transfer with consultation (alerting only) ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Call can be transferred'''
|
|----
|Held end hears music on hold or dialling tone
|
|----
|'''Call returns to transferring device if the third'''
'''Endpoint is not available'''
|
|}
|}
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.
|
|
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|
|}
=== Blind Transfer ===
{| border="1"
!Tested feature
!Result
|----
|Call can be transferred
|
|----
|Held end hears dialling tone
|
|}
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.
|
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|}
=== Blind Transfer (alerting only)===
{| border="1"
!Tested feature
!Result
|----
|Call can be transferred
|
|----
|Held end hears dialling tone
|
|}
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.
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|}
=== Broadcast Group & Waiting Queue ===
{| border="1"
!Tested feature
!Result
|----
|'''Caller can make a call to a Broadcast Group'''
|
|----
|'''Caller can make a call to a Waiting Queue'''
|
|----
|'''Announcement if nobody picks up the call'''
|
|}
== Configuration ==
===Firmware version===
All innovaphone devices use Vx build xx-xxxxx as firmware.
=== SIP - Trunk ===
=== Number Mapping ===
=== Route Settings ===
=== Media Relay ===
=== Fax ===


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

Revision as of 16:22, 3 December 2012

Innovaphone Compatibility Test Report

Summary

SIP Provider: Skype

Current test state

This product is being tested right now. The test is not yet completed.


Testing Enviroment

HFO SIP Compatibility Test 5.PNG

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. 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

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.

Tools clipart.png FIXME: 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

Tested feature Result
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.
call using g711u Nok - outbound call one way media
call using g723 Nok
call using g729 Ok
call using g722 Nok
Overlapped sending Nok
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
Fax using T.38 Not supported as written on product description, not tested
Reverse Media Negotiation Ok
CGPN can be suppressed CGPN not displayed
CLIP no screening CGPN not displayed
Long time call possible(>30 min) Not tested
External Transfer Not tested
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) Not tested
Outbound(Innovaphone -> Provider) Nok, CGPN shown as anonym

DTMF

Tested feature Result
DTMF tones sent correctly Nok via RFC-2833, only inband in RTP-Stream
DTMF tones sent correctly via SIP-Info Nok
DTMF tones received correctly Not tested

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, only with mediarelay. Without mediarelay no audio after transfer.
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.
inno1 calls sip-provider-phone. inno1 transfers to inno2.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.

Transfer with consultation (alerting only)

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

Endpoint is not available

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.
inno1 calls sip-provider-phone. inno1 transfers to inno2.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.

Blind Transfer

Tested feature Result
Call can be transferred
Held end hears dialling tone

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.
inno1 calls sip-provider-phone. inno1 transfers to inno2.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.

Blind Transfer (alerting only)

Tested feature Result
Call can be transferred
Held end hears dialling tone

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.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.

Broadcast Group & Waiting Queue

Tested feature Result
Caller can make a call to a Broadcast Group
Caller can make a call to a Waiting Queue
Announcement if nobody picks up the call

Configuration

Firmware version

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

SIP - Trunk

Number Mapping

Route Settings

Media Relay

Fax

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