Howto:HFO NGN Connect - HFO - SIP Provider

Innovaphone Compatibility Test Report

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

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 x% 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

This test verifys if the providers supports the Direct Dial In(DDI) feature. This is very important, without DDI the provider cannot be used in company enviroments. The provider offers the customer a trunk number and a phone extenion intervall.

Tested feature	Result
Inbound(Provider -> Innovaphone)	Yes
Outbound(Innovaphone -> Provider)	Yes

When configuring the IP800 it is very important to make the correct number mappings. You will find out more on this issue by scrolling to chapter "Configuration" .

DTMF

DTMF is also a must have feature for a company. DTMF is crucial for the use of a voicemail system. Currently there are two methods of transfering DTMF signals, by SIP - INFO message or encapsulated in the RTP - packet. Innovaphone supports both types of DTMF signalling.

Tested feature	Result	Comments
DTMF tones sent correctly	Yes	HFO acceps both RTP - Packets and SIP-INFO Messages for DTMF Tones
DTMF tones received correctly	Yes	HFO acceps both RTP - Packets and SIP-INFO Messages for DTMF Tones
DTMF tones audible in both directions	Yes

Hold/Retrieve

When a call is put on hold, users normally expect to hear some kind of music/announcment signalling them that they should wait. However there are two possibilities. The PBX generates the announcement or the provide generates it.

If you want to test a PBX generated announcement, then use the R - key to hold a conversation. This type of holding is tested in the Hold/Retrieve , Transfer with consultation and Transfer with consultation (alerting only) scenario.

If you want to test a provider generated announcement, then use the redial key to hold the conversation. This is used when doing a blind transfer.

Tested feature	Result
Call can be put on hold	Yes
Held end hears music on hold / announcement from PBX	No
Held end hears music on hold / announcement from provider	No
Either call can be terminated or be retrieved	Yes

Transfer with consultation

Tested feature	Result
Call can be transfered	Yes
Held end hears music on hold	No
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	No
Call returns to transferring device if the third Endpoint is not available	Yes

Blind Transfer

When using a blind transfer, the accepts the call and relays it to another callee. This is done by pressing the redial key, typing the desired phone number, followed by the OK key (IP110 & IP230) or the Enter key (IP200).

Tested feature	Result S1
Call can be transfered	Yes
Held end hears dialing tone	No

Broadcast Group & Waiting Queue

From the technical point of view, this features have been tested already. The provider must be able to switch between Music on Hold, announcements and the responding caller. The heavy load of the callswitching is done by the PBX.

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

Calling Party Number

Here we tested if the provider accepts the phone extension (DDI) and forwards the calling number correctly. Also CGPN suppresion was tested. You can enable CGPN suppresion, directly at the IP200. You can suppress your number by enabling the checkbox "Hide own Number" found under Configuration -> "Registration x" -> Preferences -> "Hide own Number".

Tested feature	Result S1
CGPN is displayed correctly	Yes
CGPN can be supressed	No

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 (see Media Relay ); 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.