Howto:Sipgate SIP Provider Compatibility Test: Difference between revisions

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** G711
** G711
** G729
** G729
** G723
** G726
** G726


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Sipgate does not support the hold feature correctly. When making a transfer the remote end will not hear a MOH/dialtone.
Sipgate does not support the hold feature correctly. When making a transfer the remote end will not hear a MOH/dialtone.


Sipgate 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]]
Sipgate has achieved 89% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]]


== 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}}-->
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== 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.


=== Basic Call ===
=== Basic Call ===
The purpose of  the ''Basic Call'' tests is to verify some standard provider features, like supported codecs and their overall voicequality. Also tested is the early media channel capabilities of the provider. Most SIP - Provider will not support early media, they will send SIP Status Messages (e.g. 404 User Not Found) instead of a Voice Stream(RTP) containing the same information. However Sipgate does support partially early media. They use this feature to send pricing information to the customer (e.g. "Only 1 cent per minute.")


{| border="1"  
{| border="1"  
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!Result S2
!Result S2
|----
|----
|call using g711a
|'''call using g711a'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|call using g711u
|'''call using g711u'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|call using g723
|call using g723
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|No
|No
|----
|----
|early media channel
|'''early media channel'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|Fax
|Fax using T.38
|No
|No
|No
|No
|----
|----
|Voice Quality OK?
|CGPN can be supressed
|Yes
|Yes
|Yes
|Yes
|----
|'''Reverse Media Negotiaton'''
|'''Yes'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|'''Yes'''
|}
|}


 
=== Direct Dial In ===
=== Dial Inward ===




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!Result S2
!Result S2
|----
|----
|Inbound(Provider -> Innovaphone)
|'''Inbound(Provider -> Innovaphone)'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|Outbound(Innovaphone -> Provider)
|'''Outbound(Innovaphone -> Provider)'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|}
|}


=== DTMF ===
=== DTMF ===
DTMF is also a must have feature for a company. DTMF is crucial for the use of a voicemail system. Currently there are two methods of transfering DTMF signals, by SIP - INFO message or encapsulated in the RTP - packet. Innovaphone supports both types of DTMF signalling. However you must pay attention at the proper configuration of your innovaphone box, since your provider will typical support just one kind of DTMF tone siganlisation.


{| border="1"  
{| border="1"  
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!Result S1
!Result S1
!Result S2
!Result S2
!Comments
|----
|----
|DTMF tones sent correctly
|'''DTMF tones sent correctly'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|Sipgate uses SIP - INFO for DTMF
|----
|DTMF tones received correctly
|Yes
|Yes
|Sipgate uses SIP - INFO for DTMF
|----
|----
|DTMF tones audible in both directions
|'''DTMF tones received correctly'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|
|}
|}


=== Hold/Retrieve ===
=== Hold/Retrieve ===
When a call is put on hold, users normally expect to hear some kind of music/announcment signalling them that they should wait. However there are two possibilities. The PBX generates the announcement or the provide generates it.


{| border="1"  
{| border="1"  
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!Result S2
!Result S2
|----
|----
|Device can put call on hold
|'''Call can be put on hold'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|Held end hears music on hold/announcement from PBX
|Held end hears music on hold / announcement from PBX
|Yes
|Yes
|Yes
|Yes
|----
|----
|Held end hears music on hold/announcement from Provider
|Held end hears music on hold / announcement from provider
|No
|No
|No
|No
|----
|Device can terminate either call and retrieve remaining call
|Yes
|Yes
|}
|}


=== Transfer with consultation===
=== Transfer with consultation===


{| border="1"  
{| border="1"  
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!Result S2
!Result S2
|----
|----
|Device can transfer call
|'''Call can be transfered'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
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|Yes
|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
|'''Yes'''
|Yes
|'''Yes'''
|}
|}


=== Transfer with consultation (alerting only)===
=== Transfer with consultation (alerting only)===
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!Result S2
!Result S2
|----
|----
|Device can transfer call
|'''Call can be transfered'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|Held end hears music on hold or dialing tone
|Held end hears music on hold or dialing tone
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|Yes
|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
|'''Yes'''
|Yes
|'''Yes'''
|}
|}


=== Blind Transfer ===
=== Blind Transfer ===
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=== Broadcast Group & Waiting Queue ===
=== Broadcast Group & Waiting Queue ===
From thee technical point of view, this features have been testede already. The provider must be able to switch between Music on Hold, announcements and the responding caller. The heavy load of the callswitching is done by the PBX.


{| border="1"  
{| border="1"  
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!Result S2
!Result S2
|----
|----
|Caller can make a call to a Broadcast Group
|'''Caller can make a call to a Broadcast Group'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|Caller can make a call to a Waiting Queue
|'''Caller can make a call to a Waiting Queue'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|----
|----
|Announcement if nobody picks up the call
|'''Announcement if nobody picks up the call'''
|Yes
|'''Yes'''
|Yes
|'''Yes'''
|}
|}


=== Calling Party Number ===
Here we tested if the provider accepts the phone extension (DDI) and forwards the calling number correctly. Also CGPN suppresion was tested. You can enable CGPN suppresion, directly at the IP200. You can suppress your number by enabling the checkbox "Hide own Number" found under
Configuration -> "Registration x" -> Preferences -> "Hide own Number".
{| border="1"
!Tested feature
!Result S1
!Result S2
|----
|CGPN is displayed correctly
|Yes
|Yes
|----
|CGPN can be supressed
|Yes
|Yes
|}




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Now the PBX and the phones are setup correctly. You should be able to make call in both directions.
Now the PBX and the phones are setup correctly. You should be able to make call in both directions.
=== DTMF ===
The only point which still has to be configured is the way outgoing DTMF signalls are sent. Sipgate accepts only DTMF signalls transmitted by SIP-INFO messages, also called Outband - signalling. You must disable the "Send DTMF Tones as RTP-DTMF" checkbox at your IP200 phone. This is also the default setting.
[[Image:Sipgate_SIP_Compatibility_Test_4.PNG‎]]




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

Latest revision as of 17:48, 3 March 2010

Innovaphone Compatibility Test Report

Summary

SIP Provider: Sipgate

  • Features:
    • Direct Dial In
    • DTMF
  • Supported Codecs by the provider
    • G711
    • G729
    • G726


The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.

Sipgate does not support the hold feature correctly. When making a transfer the remote end will not hear a MOH/dialtone.

Sipgate has achieved 89% of all possible test points. For more information on the test rating, please refer to Test Description

Current test state

The tests for this product have been completed.

Testing of this product has been finalized July 11th, 2007.

Testing Enviroment

Scenario 1 (S1)

This scenario tries to be as simple as possible. The IP - PBX and the phones are all in the same network, having public IP - adresses. The signalling channel will pass through the PBX, the media channel will go from endpoint (phone) to endpoint (provider).

CompatProviderSIP-1.JPG

Scenario 2 (S2)

This scenario demonstrates a typical office installation. The IP - PBX uses both its interfaces, one interface has a public IP adress the other interface has a private IP adress. The signalling and also the media channel will be relayed by the PBX. This kind of setup is used more often then the first scenario, because of the limited ammount of public adresses.

CompatProviderSIP-2.JPG


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 S1 Result S2
call using g711a Yes Yes
call using g711u Yes Yes
call using g723 No No
call using g729 Yes Yes
Overlapped sending No No
early media channel Yes Yes
Fax using T.38 No No
CGPN can be supressed Yes Yes
Reverse Media Negotiaton Yes Yes
Voice Quality OK? Yes Yes

Direct Dial In

This test verifys if the providers supports the Direct Dial In(DDI) feature. This is very important, without DDI the provider cannot be used in company enviroments. The provider offers the customer a trunk number and a phone extenion intervall. Inovaphone uses the SIP Prefered - Identity Header to communicate the extension to the provider. Keep in mind that Sipgate will offer you only 4 phone extensions, 0 - 2 and 9.


Tested feature Result S1 Result S2
Inbound(Provider -> Innovaphone) Yes Yes
Outbound(Innovaphone -> Provider) Yes Yes

DTMF

Tested feature Result S1 Result S2
DTMF tones sent correctly Yes Yes
DTMF tones received correctly Yes Yes

Hold/Retrieve

Tested feature Result S1 Result S2
Call can be put on hold Yes Yes
Held end hears music on hold / announcement from PBX Yes Yes
Held end hears music on hold / announcement from provider No No

Transfer with consultation

Tested feature Result S1 Result S2
Call can be transfered Yes Yes
Held end hears music on hold Yes Yes
Call returns to transferring device if the third

Endpoint is not available

Yes Yes

Transfer with consultation (alerting only)

Tested feature Result S1 Result S2
Call can be transfered Yes Yes
Held end hears music on hold or dialing tone Yes Yes
Call returns to transferring device if the third

Endpoint is not available

Yes Yes

Blind Transfer

Tested feature Result S1 Result S2
Device can transfer call Yes Yes
Held end hears dialing tone No No


Broadcast Group & Waiting Queue

Tested feature Result S1 Result S2
Caller can make a call to a Broadcast Group Yes Yes
Caller can make a call to a Waiting Queue Yes Yes
Announcement if nobody picks up the call Yes Yes


Configuration

SIP - Trunk

First of all the SIP Trunk must be configured. Here an example of our Sipgate - Trunk.


Sipgate SIP Compatibility Test 1.PNG

Number Mapping

The complicated part on this issue is the correct mapping of the outgoing and incoming numbers. Sipgate requires an E.164 number format, so the outgoing number must look something like this: 49703.. . However do not map your outgoing number to the format 0049703.. . Sipgate will accept outgoing calls but incoming calls for the number 00497031.. will not be forwarded correctly. The result is that you can call everybody, but they can't calll you back.

Sipgate SIP Compatibility Test 2.PNG


Sipgate Account Settings

In order for the number mapping to work correctly you must make some minor adjustments to your Sipgate Account. To do this, go to "Einstellungen" (Settings) -> "Telefonie" and change the settings of the paragraph "Absenderrufnummer setzen" to the value "setzt das Engerät".


Sipgate SIP Compatibility Test 5.PNG


For additional help on configuring a SIP trunk with Sipgate, please refer to: [www.sipgate.de/user/trunking.php]


Enblock Sending

Because Sipgate, as most SIP - Providers too, doesn't support overlap sending, you must enable the blockwise sending of the phone number. You can do this by making some minor changes in the automatically generated Routes.


Sipgate SIP Compatibility Test 3.PNG


Media Relay (Scenario 2 only)

By enabling Media Relay on the PBX, The IP800 will work as an RTP - Proxy, so all RTP Streams will travel through the IP800. This mode poses a much greater load on the PBX, so the number of concurrent calls will be heavy limited.


Toplink SIP Compatibility Test 5.PNG


You may wonder about the usefullness of putting the localhost address as a private network. However you must insert this entry, if the IP800 SIP GW registers at the PBX on localhost interface.


Now the PBX and the phones are setup correctly. You should be able to make call in both directions.

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