Howto:HFO NGN Connect - HFO - SIP Provider

Innovaphone Compatibility Test Report

Summary

SIP Provider: HFO

• Features:

• • Direct Dial In
  • Fax over IP (T.38)
  • DTMF

• Supported Codecs by the provider
  • G711
  • G722
  • T.38 UDP

Current test state

This product is being tested right now. The test is not yet completed.

<internal> Beim Abschluss des Tests (egal ob gut, schlecht oder abgebrochen) bitte Nachricht an ckl! </internal>

Testing Environment

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

There are 3 major configuration variants of the SIP trunk, which one is used depends on the test results. The variants are:

• the SIP trunk is configured without Media Relay and without exclusive coder. This is the case if all tests were successful
• the SIP trunk is configured with Media Relay and STUN but without exclusive coder. This is the case when the test for "NAT Traversal" fails
• the SIP trunk is configured with Media Relay and exclusive coder. This is the case when the test for "Reverse Media Negotiation" fails

The test scenario should describe which SIP trunk configuration is needed.

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	Nok
Redundancy	Ok
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 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.

Fax

Innovaphone Compatibility Test Report-old-test

Summary

SIP Provider: HFO

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

HFO does not support the remote hold feature. When making a blind transfer(using redial key) the remote end will not get a MOH/dialtone from the provider.

HFO has achieved 89% of all possible test points. For more information on the test rating, please refer to Test Description

• Features:

• • Direct Dial In
  • DTMF

• Supported Codecs by the provider
  • G711
  • G729

Current test state

The tests for this product have been completed.

Testing of this product has been finalized October 22th, 2007.

Testing Enviroment

Scenario NAT

This scenario describes a setup where the PBX and phones are in a private network. The IP800 must 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.

Basic Call

Tested feature	Result
call using g711a	Yes
call using g711u	Yes
call using g723	No
call using g729	Yes
Overlapped sending	No
early media channel	Yes
Fax using T.38	No
CGPN can be supressed	Yes
Reverse Media Negotiaton	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 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	No

Transfer with consultation

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

Blind Transfer

Tested feature	Result
Call can be transfered	Yes
Held end hears dialing tone	No

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

Configuration

General Information

Basic Provider Infomation: HFO

• described in Mantis Case: 16505

Firmware version

• IP800: 6.00 dvl-sr2 IP800[07-60600.58]
• IP22: 6.00 dvl-sr1 IP230[07-60600.58]
• IP200: 6.00 dvl-sr1 IP230[07-60600.58]
• IP230: 6.00 dvl-sr1 IP230[07-60600.58]

SIP - Trunk

First of all the SIP Trunk must be configured. Here an example of our HFO - Trunk. If you want that the Gateway should act as Medialrelay, you must exchange the Gatekeeper Address (blue marking) to an private network address; for example 127.0.0.1.

HFO awaits in the From Header the complete Calling Party Number(CGPN). The default innovaphone setting is to not send the complete CGPN in the FROM - Header, but in the Preffered Identity Header. Change the setting From Header: to CGPN in user part of URI.

Number Mapping

The complicated part on this issue is the correct mapping of the outgoing and incoming numbers.

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.

The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay suplementary services, like Hold over the SIP Trunk. If this checkbox is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.

Media Relay

By enabling Media Relay on the PBX, The IP800 will work as an RTP - Proxy, so all RTP Streams will travel through the IP800. This mode poses a much greater load on the PBX, so the number of concurrent calls will be heavy limited.

You may wonder about the usefullness of putting the localhost address as a private network. However you must insert this entry, if the IP800 SIP GW registers at the PBX on localhost interface.

Now the PBX and the phones are setup correctly. You should be able to make call in both directions and send and receive fax messages.