Howto:Perustele Oy SIP Provider Compatibility Test

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Innovaphone Compatibility Test Report

Contents

Summary

SIP Provider: Perustele Oy

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

That being said, the provider has achieved 83,90% of all possible test points. For more information on the test rating, please refer to Test Description

During the tests we had some issues regarding Media Negotiation, so to work proper we must set Exclusive Codec and Media-Relay ON in the SIP Trunk.

More information could be found at Known Issues


  • Features:
    • Direct Dial In
    • DTMF
  • Supported Codecs by the provider
    • G711
    • G729
    • G723
    • G726

Current test state

Image:Recprod.PNG The tests for this product have been completed and it has been approved as a recommended product (Certification document).

Testing of this product has been finalized March 4th, 2012.

Testing Enviroment

Image: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 with Media Relay but without exclusive coder. This is the case when the test for "NAT Traversal" fails

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 OK
call using g729 OK
call using g722 NOK
Overlapped sending NOK
early media channel OK
Fax using T.38 NOK
Reverse Media Negotiation NOK
CGPN can be suppressed NOK
CLIP no screening NOK
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection NOK
Redundancy OK
SIP over TCP NOK
Voice Quality OK? OK

Direct Dial In

Tested feature Result
Inbound(Provider -> Innovaphone) OK
Outbound(Innovaphone -> Provider) OK
Loop In call(Innovaphone -> Provider -> Innovaphone) 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

"*" - Blind Transfer only works if we set exclusive coder on the SIP Interface.

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

CFNR / Blind Transfer (alerting only)

Tested feature Result
Call can be transferred or forward 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. 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 Hotfix20 as firmware.

SIP - Trunk

Image:Perustele_Oy_SIP_Provider_Compatibility_Test_1.png

Number Mapping

Image:Perustele_Oy_SIP_Provider_Compatibility_Test_2.png

Route Settings

Image:Perustele_Oy_SIP_Provider_Compatibility_Test_3.png

Redundancy

The solution for redundancy, it is not straight forward.

As a workaround in cases with forwarding to SIP URI Port can using the domain name instead of IP. In this case Provider can configure DNS server to send the requests to slave PBX if the master PBX is not available.

Known Issues

  • Reverse Media Negotiation - This feature it's not supported, due that fact it's necessary to set Exclusive Codec and Media-Relay options in the SIP Trunk configuration.
  • No T.38 - During the SIP negotation T.38 was accepted however the fax transmission fails.
  • Interworking/QSIG could cause issues, reported by costumer when performing Hold and using Interworking/QSIG flag on the routes the media change offer sent to the Provider was rejected. So the solution should be not use this option as shown in the configuration above.
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