Howto:Net2phone SIP Provider Compatibility Test

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Innovaphone Compatibility Test Report

Contents

Summary

SIP Provider: Net2phone

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

It's required to use Media-relay and Exclusive Codec on the SIP interface in order to work properly, more information why could be found bellow in Known Issues bellow.

The coder G711alaw it's not supported however since G711ulaw is it we only require one of both to pass the tests.

That being said, the provider has achieved 82,70% of all possible test points (110 of 133). For more information on the test rating, please refer to Test Description


  • Features:
    • Direct Dial In
    • Fax over IP (T.38)
    • DTMF
    • CLIP No Screening
  • Supported Codecs by the provider
    • G711u
    • G729
    • G723
    • T.38 UDP

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 on October 15th 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 and exclusive codec, no STUN required.

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 NOK
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 OK
Reverse Media Negotiation NOK
CGPN can be suppressed NOK
CLIP no screening OK
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

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/Alert 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

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

Blind Transfer (alerting only)

Tested feature Result
Call can be transferred 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 hotfix 16 build 9.061101 as firmware.

SIP - Trunk

Here's the configuration of the SIP gateway interface.

Image:Net2phone_SIP_Provider_Compatibility_Test_1.png

Number Mapping

The CDPN it's equal to the account number and not the public DDI.

Image:Net2phone_SIP_Provider_Compatibility_Test_2.png

Route Settings

Force Enblock Setting it's required for outgoing calls. QSIG/SIP Interworking setting it's recommended.

Image:Net2phone_SIP_Provider_Compatibility_Test_3.png

Redundancy

From an end user point of view, the Net2phone platform will route the inbound call to the PBX that is currently registered with the SIP proxy server (i.e. the most recent PBX that is registered with Net2phone). The customer can setup 2 SIP trunks with Net2phone from 2 different PBXs using the same service account but the inbound calls will *only* route to the most recent PBX that is registered. The inbound calls are not going to ring simultaneously on both the Primary and Standby PBX. That being said, if the customer experiences an outage on their Primary PBX and the PBX looses its SIP registration, the inbound calls will route to the Standby-PBX (once the SIP registration from the Primary PBX drops - it will take a few minutes for the SIP registration from the Primary PBX to drop on Net2phone side).

Known Issues

  • No NAT Detection makes Media-Relay option necessary
Net2phone platform will send the RTP (media traffic) to the media gateway IP address and port that was negotiated in the initial SIP INVITE
to our Platform (whatever IP and port was included in the SDP). In the event that a customer is transferring a call and the SDP is changing 
then it would be best if the user sets media-relay in their PBX so that the media will continue to be sent to the media IP and port that was 
negotiated in the initial SIP INVITE. This would be the best way to handle these types of internal media stream changes. If this is not possible
then we would need to see a SIP re-INVITE from the customer with the new media IP address and port in their SDP.
  • No Reverse Media Negotiation makes Exclusive Coder option necessary
Net2phone platform doesn’t support a SIP Offer without an SDP. The SDP always
has to be included in the SIP INVITE. Otherwise we will reply back with a
SIP 415 Media Type missing
  • G711a not supported Codec
Just to clarify further regarding the supported codecs, the Net2phone IDT platform
used for the SIP trunking service only *supports G711 u-law, G729 and G723
codecs*. Unfortunately G711a-law codec is not supported on this Platform.
From Net2phone side, we will negotiate the calls with our termination providers based on what codecs
were offered to us on the origination party (customer) side. So if the
customer only offers G711u-law codec on the call, we are going to negotiate
the call with our termination provider using G711u codec. Generally
speaking, we don’t recommend that a customer only offers one specific codec
in the SIP INVITE to us (G711 for example) due to the fact that not all
termination providers support G711 codec for bandwidth reasons. We
recommend that the customer setup a preferred codec list on their switch
instead and offer multiple codecs in their SIP INVITE to us (G711u, G729).
If our termination provider supports G711u codec, we will negotiate the
call using G711u. If our termination provider doesn’t support G711u, then
the call will negotiate using G729 codec.
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