Howto:QSC IPfonie extended - SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
The tests for this product have been completed.
Summary
SIP Provider: QSC
- Features:
- Direct Dial In
- DTMF
- Supported Codecs by the provider
- G711
- G729
- G723
- T.38
The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.
The configuration on innovaphone side is simple, since the built-in SIP-GW can be used to connect to the provider.
QSC 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.
Since only minor problems have arizen during the tests, it is not possible to make a statement on the QSC support quality.
QSC has achieved x% of all possible test points. For more information on the test rating, please refer to Test Description
Current test state
The tests for this product have been completed.
Testing of this product has been finalized November 30th, 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
Basic Call
| Tested feature | Result |
|---|---|
| call using g711a | Yes |
| call using g711u | Yes |
| call using g723 | Yes |
| call using g729 | Yes |
| Overlapped sending | No |
| early media channel | Yes |
| Fax using T.38 | Yes |
| CGPN can be supressed | No |
| 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 |
| DTMF tones audible in both directions | 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 |
| 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 | 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 S1 |
|---|---|
| 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: QSC
- described in Mantis Case: 22169
Firmware version
- IP800: 6.00 dvl-sr2 IP800[07-60600.80]
- 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 QSC - 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. QSC 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 QSC, 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.
Fax
The FAX was connected via a IP22 to the IP800. When configuring the IP22 you must keep in mind that you will use the analog interface for fax communication. Thats why the T.38 codec must be enabled.
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.