Howto:Perustele Oy SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
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
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
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
Number Mapping
Route Settings
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.