Howto:HFO NGN Connect - HFO - SIP Provider

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Innovaphone Compatibility Test Report

Summary

SIP Provider: HFO

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

Only if the Mobility-feature or WebRTC is desired by the customer, you must enable Media-Relay on the SIP-Trunk. Opposed to a SIP trunk not needing Media-Relay, the transport of all RTP packets by the gateway will result in a higher CPU load for a call. As a result, the amount of concurrent calls is considerably lower compared to a SIP-Provider that doesn't require Media-Relay.

The provider doesn't support the G.729 coder, this might be an issue in locations with low-bandwidth Internet connection.

The PBX is connected to the provider through the public Internet, the tested product "HFO NGN Connect" comes without a dedicated Internet Access Device. According to the provider the amount of concurrent calls is limited when ordering the trunk, currently max. 220 concurrent calls are possible.

The provider offers an additional SIP-trunk product called HFO NGN Business. This product comes with a dedicated Internet Access, used only for telephony. According to HFO, it uses the same SIP server as the tested product "HFO NGN Connect". As a result, the configuration on innovaphone side should be equal. For more details on the product "HFO NGN Business", please contact the provider directly.

NAT - Detection doesn't work for all private networks, see Known Problems for more information.

That being said, the provider has achieved 92% of all possible test points. 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
    • G711
    • G722
    • T.38 UDP

Current test state

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 June 23th, 2015.

Testing Environment

SIPProviderTestTopology1.PNG

This scenario describes a setup where the PBX and phones are in a private network.


  • the SIP trunk is configured without Media Relay and without exclusive coder. Media-Relay must be only activated if Mobility is used.


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
SIP over TLS(SIPS) Nok
SIP over TCP Nok
SRTP Nok
call using g711a Ok
call using g711u Ok
call using g729 Nok
call using g722 Ok
Overlapped sending Nok
early media channel Ok
Fax using T.38 Ok
T.38 Transcoding by the provider Ok
Reverse Media Negotiation Ok
CGPN can be suppressed Ok
CLIP no screening Ok
Long time call possible(>30 min) Ok
External Transfer Ok
NAT Detection Ok*
Redundancy Ok
Voice Quality OK? Ok
* NAT - Detection doesn't work for all private networks, see section  Known Problems for more information.

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 via RTP-events(RFC 2833) Ok
DTMF tones sent correctly via SIP-Info Nok
DTMF tones received correctly via RTP-events(RFC 2833) 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 PSTN-phone. Ok Ok
inno1 calls PSTN-phone. inno1 transfers to inno2. Ok Ok
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. Ok Ok
PSTN-phone calls inno1. inno1 transfers to inno2. Ok Ok
PSTN-phone calls inno1. PSTN-phone transfers to inno2. Ok Ok
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. 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 PSTN-phone. Ok Ok
inno1 calls PSTN-phone. inno1 transfers to inno2. Ok Ok
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. Ok Ok
PSTN-phone calls inno1. inno1 transfers to inno2. Ok Ok
PSTN-phone calls inno1. PSTN-phone transfers to inno2. Ok Ok
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. 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 PSTN-phone. Ok
inno1 calls PSTN-phone. inno1 transfers to inno2. Ok
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. Ok
PSTN-phone calls inno1. inno1 transfers to inno2. Ok
PSTN-phone calls inno1. PSTN-phone transfers to inno2. Ok
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. Ok

CFU / CFB Transfer

Tested feature Result
Call can be forwarded Ok
Held end hears dialling tone Ok

CFNR / Blind Transfer (alerting only)

Tested feature Result
Call can be transferred or forwarded 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 PSTN-phone. Ok
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. Ok
PSTN-phone calls inno1. inno1 transfers to inno2. Ok
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. 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 V11r2 SR1 as firmware.

SIP - Trunk

First of all the SIP Trunk must be configured. Here an example of our HFO - Trunk. If the Mobility-feature is desired by the customer, you must enable Media-Relay on the SIP-Trunk.

HFO SIP.PNG

Number Mapping

The provider uses a CDPN in national format for incoming call. Therefore the SIP-interface maps must be configured accordingly.

HFO SIP Compatibility Test 2.PNG


Route Settings

Because HFO, 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 enabling Force enblock in the automatically generated Routes.

HFO SIP Compatibility Test 3.PNG

If there is a problem with the number format, make sure to read also Howto:Create_ClipNoScreening_Maps_for_SIP_Interfaces

Fax

The provider supports T.38, make sure to enable it on the SIP-interface and also on all analogue - interfaces used for fax-machines .

Known Problems

HFO doesn't support NAT Traversal for clients using private addresses in 172.16.0.0/24 to 172.31.0.0/24 networks. In case that this private networks are used, STUN must be used at the RTP endpoints. In order for NAT-Traversal using STUN to work, the NAT-Router must support full-cone or restricted NAT.