Howto:QSC VoIP connect - 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
Technically, QSC VoIP Connect meets all requirements for a recommended SIP trunk. However, VoIP connect is a wholesale SIP trunk product. It implements phone calls for many customers (i.e. many subscriber numbers) on a single SIP trunk (a.k.a. fat pipe). The operator of such a trunk has to meet the legal obligations of a telecommunication carrier, as imposed by the laws in the respective country. As a result, users of this product need to be prepared to fulfil all such requirements. If you plan to use the VoIP connect trunk, we recommend contacting innovaphone pre-sales for discussing your concept/planned installation.
For customers who do not want to take over carrier responsibilities, we recommend using QSC's IPfonie extended product instead.
Current test state
The tests for this product have been completed.
Testing of this product has been finalized November 19th, 2009.
Testing Enviroment
In this scenario the PBX is in the public network. The registered phones are alternatively in private or public networks. STUN is not used, the SIP provider is responsible for handling SIP messages with private IP addresses. Moreover mediarelay is not configured, resulting in RTP streams not passing the PBX but going directly between the SIP provider and the used phone.
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 | Yes |
call using g729 | Yes |
Overlapped sending | No |
early media channel | Yes |
Fax using T.38 | Yes |
CGPN can be suppressed | Yes (with special flag) |
CLIP no screening | No |
Reverse Media Negotiaton | Yes |
Long time call possible(>30 min) | 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 |
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 | Yes |
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
Firmware version
All innovaphone devices use V8 firmware.
SIP - Trunk
First of all the SIP Trunk must be configured. Since QSC authenticates a user account only by IP - Address, you cannot use the normal SIP-Gateway object. You need to configure a GW without registration and route the calls to the PBX.
Number Mapping
Route Settings
Because QSC, as most SIP - Providers too, doesn't support overlap sending, you must enable the block-wise sending of the phone number. You can do this by enabling Force enblock in your routes.
The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this check-box is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.
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