Howto:Sipgate SIP Provider Compatibility Test

Innovaphone Compatibility Test Report

Summary

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

Supported Codecs

Codec	Applies
G711	Yes
G729	Yes
G723	Yes
G726	Yes
GSM	Yes
T.38 UDP	Yes
G722	No

Testing Enviroment

Scenario 1 (S1)

This scenario tries to be as simple as possible. The IP - PBX and the phones are all in the same network, having public IP - adresses. The signalling channel will pass through the PBX, the media channel will go from endpoint (phone) to endpoint (provider).

Scenario 2 (S2)

This scenario demonstrates a typical office installation. The IP - PBX uses both its interfaces, one interface has a public IP adress the other interface has a private IP adress. The signalling and also the media channel will be relayed by the PBX. This kind of setup is used more often then the first scenario, because of the limited ammount of public adresses.

Test Results

Basic Call

The purpose of the Basic Call tests is to verify some standard provider features, like supported codecs and their overall voicequality. Also tested is the early media channel capabilities of the provider. Most SIP - Provider will not support early media, they will send SIP Status Messages (e.g. 404 User Not Found) instead of a Voice Stream(RTP) containing the same information. However Sipgate does support partially early media. They use this feature to send pricing information to the customer (e.g. "Only 1 cent per minute.")

Tested feature	Result S1	Result S2
call using g711a	Yes	Yes
call using g711u	Yes	Yes
call using g723	No	No
call using g729	Yes	Yes
Overlapped sending	No	No
early media channel	Yes	Yes
Fax	No	No
Voice Quality OK?	Yes	Yes

Dial Inward

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. Inovaphone uses the SIP Prefered - Identity Header to communicate the extension to the provider. Keep in mind that Sipgate will offer you only 4 phone extensions, 0 - 2 and 9.

Tested feature	Result S1	Result S2
Inbound(Provider -> Innovaphone)	Yes	Yes
Outbound(Innovaphone -> Provider)	Yes	Yes

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. However you must pay attention at the proper configuration of your innovaphone box, since your provider will typical support just one kind of DTMF tone siganlisation.

Tested feature	Result S1	Result S2	Comments
DTMF tones sent correctly	Yes	Yes	Sipgate uses SIP - INFO for DTMF
DTMF tones received correctly	Yes	Yes	Sipgate uses SIP - INFO for DTMF
DTMF tones audible in both directions	Yes	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.

Tested feature	Result S1	Result S2
Device can put call on hold	Yes	Yes
Held end hears music on hold/announcement from PBX	Yes	Yes
Held end hears music on hold/announcement from Provider	No	No
Device can terminate either call and retrieve remaining call	Yes	Yes

Transfer with consultation

Tested feature	Result S1	Result S2
Device can transfer call	Yes	Yes
Held end hears music on hold	Yes	Yes
Call returns to transferring device if the third Endpoint is not available	Yes	Yes

Transfer with consultation (alerting only)

Tested feature	Result S1	Result S2
Device can transfer call	Yes	Yes
Held end hears music on hold or dialing tone	Yes	Yes
Call returns to transferring device if the third Endpoint is not available	Yes	Yes

Blind Transfer

Tested feature	Result S1	Result S2
Device can transfer call	Yes	Yes
Held end hears dialing tone	No	No

Broadcast Group & Waiting Queue

From thee technical point of view, this features have been testede 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 S1	Result S2
Caller can make a call to a Broadcast Group	Yes	Yes
Caller can make a call to a Waiting Queue	Yes	Yes
Announcement if nobody picks up the call	Yes	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	Result S2
CGPN is displayed correctly	Yes	Yes
CGPN can be supressed	Yes	Yes

Configuration

SIP - Trunk

First of all the SIP Trunk must be configured. Here an example of our Sipgate - Trunk.

Number Mapping

The complicated part on this issue is the correct mapping of the outgoing and incoming numbers. Sipgate requires an E.164 number format, so the outgoing number must look something like this: 49703.. . However do not map your outgoing number to the format 0049703.. . Sipgate will accept outgoing calls but incoming calls for the number 00497031.. will not be forwarded correctly. The result is that you can call everybody, but they can't calll you back.

Sipgate Account Settings

In order for the number mapping to work correctly you must make some minor adjustments to your Sipgate Account. To do this, go to "Einstellungen" (Settings) -> "Telefonie" and change the settings of the paragraph "Absenderrufnummer setzen" to the value "setzt das Engerät".

For additional help on configuring a SIP trunk with Sipgate, please refer to: [www.sipgate.de/user/trunking.php]

Enblock Sending

Because Sipgate, 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 making some minor changes in the automatically generated Routes.

Media Relay (Scenario 2 only)

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.

DTMF

The only point which still has to be configured is the way outgoing DTMF signalls are sent. Sipgate accepts only DTMF signalls transmitted by SIP-INFO messages, also called Outband - signalling. You must disable the "Send DTMF Tones as RTP-DTMF" checkbox at your IP200 phone. This is also the default setting.