Howto:Uni-tel SIP Provider Compatibility Test

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Summary

SIP Provider: Uni-tel A/S

The provider has achieved 93,24% of all possible test points (124 on total of 133 Points). For more information on the test rating, please refer to Test Description


  • Features:
    • Direct Dial In
    • DTMF
    • Redundancy Mechanism


  • Supported Codecs by the provider
    • G711 a/u

Current test state

Recprod.PNG 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 in 28th August, 2012.

Testing Enviroment

Scenario NAT

HFO SIP Compatibility Test 5.PNG

This scenario describes a setup where the PBX and phones are in a private network.

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 No
Call using g722 No
Overlapped sending No
Early media channel Yes
Fax using T.38 Yes
CGPN can be suppressed Yes
CLIP no screening Yes (only for Danish numbers)
Reverse Media Negotiation Yes
Long time call possible Yes
External Transfer Yes
NAT Detection Yes
Voice Quality OK? Yes
Redundancy Mechanism Yes
SIP over TCP 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 sent correctly via SIP-Info 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 transferred Yes
Held end hears music on hold Yes

Transfer with consultation (alerting only)

Tested feature Result
Call can be transferred Yes
Held end hears music on hold or dialling tone Yes
Call returns to transferring device if the third Endpoint is not available Yes

Blind Transfer

Tested feature Result
Call can be transferred Yes
Held end hears dialling 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 V9hf15 as firmware.

SIP - Trunk

Here's the configuration of the SIP gateway interface.

Uni-tel SIP Provider Compatibility Test 1.png

No need to use STUN, so It's optional.

Fixed AOR should not be used to send different CPGN since only "FROM" header it's used for CGPN and not Prefered-ID field on SIP Invite.

Number Mapping

Uni-tel SIP Provider Compatibility Test 2.png

Route Settings

Uni-tel, as most SIP Providers do, doesn't support overlap sending. You must enable the enblock sending of the phone number. You can do this by enabling Force enblock in the appropriate outgoing Route.

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.

Uni-tel SIP Provider Compatibility Test 3.png

Redundancy

Redundancy could be done by DNS SRV.

When resolving the proxy name the provider may give us several IP address in the SRV record. We select one address and keep the others as fallback. Methods used to select one address of several returned addresses are also used for load balancing. Whether load balancing and/or redundancy is implemented only depends on priority values of the returned addresses.

Known Issues