Howto:VoIP-Telecom - SIP Provider Compatibility Test: Difference between revisions

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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.
{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|No
|----
|Overlapped sending
|No
|----
|'''early media channel'''
|'''Yes'''
|----
|Fax using T.38
|No
|----
|CGPN can be suppressed
|No
|----
|CLIP no screening
|No
|----
|'''Reverse Media Negotiation'''
|'''Yes'''
|----
|'''Long time call possible'''
|'''Yes'''
|----
|'''External Transfer'''
|'''Yes'''
|----
|Voice Quality OK?
|Yes
|}
=== Direct Dial In ===
{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}
=== DTMF ===
{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones sent correctly via SIP-Info'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}
=== Hold/Retrieve ===
{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}
=== Transfer with consultation ===
{| border="1"
!Tested feature
!Result
|----
|'''Call can be transferred'''
|'''Yes'''
|----
|Held end hears music on hold
|Yes
|----
|'''Call returns to transferring device if the third Endpoint is not available'''
|'''Yes'''
|}
=== Transfer with consultation (alerting only) ===
{| border="1"
!Tested feature
!Result
|----
|'''Call can be transferred'''
|'''Yes'''
|----
|Held end hears music on hold or dialling tone
|Yes
|----
|'''Call returns to transferring device if the third Endpoint is not available'''
|'''Yes'''
|}
=== Blind Transfer ===
{| border="1"
!Tested feature
!Result
|----
|Call can be transferred
|Yes
|----
|Held end hears dialling tone
|Yes
|}
=== Broadcast Group & Waiting Queue ===
{| border="1"
!Tested feature
!Result
|----
|'''Caller can make a call to a Broadcast Group'''
|'''Yes'''
|----
|'''Caller can make a call to a Waiting Queue'''
|'''Yes'''
|----
|'''Announcement if nobody picks up the call'''
|'''Yes'''
|}


[[Image:Voip-telecom_SIP_Compatibility_Test_2.png]]
[[Image:Voip-telecom_SIP_Compatibility_Test_2.png]]

Revision as of 08:49, 28 July 2011

Voip-Telecom innovaphone Compatibility Test Report

Summary

SIP Provider: VoIP-Telecom

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

That beeing said, the provider has achieved 97,5% of all possible test points. For more information on the test rating, please refer to Test Description

The Provider have NAT Detection system so we don't need to use STUN Server to work with. It's recommended to use media-relay to guarantee that there is no problem with media in some type of calls (like external transfers).

It's required by the provider that user send 0 as prefix before the full number. Ex: 0 + 00497031730090.


  • Features:
    • Direct Dial In
    • Fax over IP (T.38)
    • DTMF
    • NAT Detection
    • Suppress the Number if Requested.


  • Supported Codecs by the provider
    • G711
    • G729
    • G723
    • T.38 UDP

Current test state

The tests for this product have been completed.

Testing of this product has been finalized July 22, 2011.

Testing Enviroment

Scenario NAT

HFO SIP Compatibility Test 5.PNG

This scenario describes a setup where the PBX and phones are in a private network. The IP800 should use a stun server, in order to send correct SIP - messages. The IP800 works as media relay, all RTP - streams go through the PBX.

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.

Tested feature Result
call using g711a Yes
call using g711u Yes
call using g723 No
call using g729 No
Overlapped sending No
early media channel Yes
Fax using T.38 No
CGPN can be suppressed No
CLIP no screening No
Reverse Media Negotiation Yes
Long time call possible Yes
External Transfer Yes
Voice Quality OK? Yes

Direct Dial In

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

DTMF

Tested feature Result
DTMF tones sent correctly Yes
DTMF tones sent correctly via SIP-Info Yes
DTMF tones received correctly Yes

Hold/Retrieve

Tested feature Result
Call can be put on hold Yes
Held end hears music on hold / announcement from PBX Yes
Held end hears music on hold / announcement from provider Yes

Transfer with consultation

Tested feature Result
Call can be transferred Yes
Held end hears music on hold Yes
Call returns to transferring device if the third Endpoint is not available Yes

Transfer with consultation (alerting only)

Tested feature Result
Call can be transferred Yes
Held end hears music on hold or dialling tone Yes
Call returns to transferring device if the third Endpoint is not available Yes

Blind Transfer

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

Broadcast Group & Waiting Queue

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


Voip-telecom SIP Compatibility Test 2.png


Configuration

Firmware version

All innovaphone devices use V9hf1 build 90600.01 as firmware.

SIP - Trunk

Voip-telecom SIP Compatibility Test 3.png

Number Mapping

Voip-telecom SIP Compatibility Test 4.png


We should add 0 on CDPN Out maps so users don't need to dial extra 0 to the Provider.

Route Settings

Since the Provider support Overlap sending there is no need to use "force enblock" option in the routes. We recommend the use of "Interworking QSIG/SIP" option in the routes.

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