Howto:QSC IPfonie extended - SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
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 has achieved 98% 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 and it has been approved as a recommended product (Certification document).
Testing of this product has been finalized March 13th, 2010.
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 | |
| call using g711u | |
| call using g723 | |
| call using g729 | |
| call using g722 | |
| Overlapped sending | |
| early media channel | |
| Fax using T.38 | |
| Reverse Media Negotiation | |
| CGPN can be suppressed | |
| CLIP no screening | |
| Long time call possible(>30 min) | |
| External Transfer | |
| NAT Detection | |
| Redundancy | |
| SIP over TCP | |
| Voice Quality OK? |
Direct Dial In
| Tested feature | Result |
|---|---|
| Inbound(Provider -> Innovaphone) | |
| Outbound(Innovaphone -> Provider) |
DTMF
| Tested feature | Result |
|---|---|
| DTMF tones sent correctly | |
| DTMF tones sent correctly via SIP-Info | |
| DTMF tones received correctly |
Hold/Retrieve
| Tested feature | Result |
|---|---|
| Call can be put on hold | |
| Held end hears music on hold / announcement from PBX |
Transfer with consultation
| Tested feature | Result |
|---|---|
| Call can be transferred | |
| Held end hears music on hold |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? | MoH Ok? |
|---|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | ||
| inno1 calls inno2. inno1 transfers to sip-provider-phone. | ||
| inno1 calls sip-provider-phone. inno1 transfers to inno2. | ||
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | ||
| sip-provider-phone calls inno1. inno1 transfers to inno2. | ||
| sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. |
Transfer with consultation (alerting only)
| Tested feature | Result |
|---|---|
| Call can be transferred | |
| Held end hears music on hold or dialling tone | |
| Call returns to transferring device if the third
Endpoint is not available |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? | MoH Ok? |
|---|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | ||
| inno1 calls inno2. inno1 transfers to sip-provider-phone. | ||
| inno1 calls sip-provider-phone. inno1 transfers to inno2. | ||
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | ||
| sip-provider-phone calls inno1. inno1 transfers to inno2. | ||
| sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. |
Blind Transfer
| Tested feature | Result |
|---|---|
| Call can be transferred | |
| Held end hears dialling tone |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? |
|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | |
| inno1 calls inno2. inno1 transfers to sip-provider-phone. | |
| inno1 calls sip-provider-phone. inno1 transfers to inno2. | |
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | |
| sip-provider-phone calls inno1. inno1 transfers to inno2. | |
| sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. |
Blind Transfer (alerting only)
| Tested feature | Result |
|---|---|
| Call can be transferred | |
| Held end hears dialling tone |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? |
|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | |
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | |
| sip-provider-phone calls inno1. inno1 transfers to inno2. |
Broadcast Group & Waiting Queue
| Tested feature | Result |
|---|---|
| Caller can make a call to a Broadcast Group | |
| Caller can make a call to a Waiting Queue | |
| Announcement if nobody picks up the call |
Configuration
Firmware version
- IP800: 8.00 dvl IP800[09-80394]
- IP302: 8.00 dvl IP800[09-80394]
- IP200: 8.00 dvl IP800[09-80394]
- IP230: 8.00 dvl IP800[09-80394]
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. The tests described in this article, were succesfull without using Media Relay. If you run into problems(media, one way audio), you can still activate the Media Relay checkbox at the SIP interface.
Fax
The FAX was connected via a IP302 to the IP800. When configuring the IP302 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.
CLIR
To suppress the CGPN for outbound calls a config line option must be activated. Please enter the following lines in your browser:
http://PBX-IP-address/!config add SIP /pai http://PBX-IP-address/!config write http://PBX-IP-address/!config activate