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Revision as of 15:26, 12 June 2015
Innovaphone Compatibility Test Report
Summary
SIP Provider: HFO
- Features:
- Direct Dial In
- Fax over IP (T.38)
- DTMF
- Supported Codecs by the provider
- G711
- G722
- T.38 UDP
Current test state
This product is being tested right now. The test is not yet completed.
<internal>
Beim Abschluss des Tests (egal ob gut, schlecht oder abgebrochen) bitte Nachricht an ckl!
</internal>
Testing Environment
This scenario describes a setup where the PBX and phones are in a private network.
There are 3 major configuration variants of the SIP trunk, which one is used depends on the test results. The variants are:
- the SIP trunk is configured without Media Relay and without exclusive coder. This is the case if all tests were successful
- the SIP trunk is configured with Media Relay and STUN but without exclusive coder. This is the case when the test for "NAT Traversal" fails
- the SIP trunk is configured with Media Relay and exclusive coder. This is the case when the test for "Reverse Media Negotiation" fails
The test scenario should describe which SIP trunk configuration is needed.
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 |
---|---|
SIP over TLS(SIPS) | Nok |
SIP over TCP | Nok |
SRTP | Nok |
call using g711a | Ok |
call using g711u | Ok |
call using g729 | Nok |
call using g722 | Ok |
Overlapped sending | Nok |
early media channel | Ok |
Fax using T.38 | Ok |
T.38 Transcoding by the provider | Ok |
Reverse Media Negotiation | Ok |
CGPN can be suppressed | Ok |
CLIP no screening | Ok |
Long time call possible(>30 min) | Ok |
External Transfer | Ok |
NAT Detection | Nok |
Redundancy | Nok, incoming call has no audio |
Voice Quality OK? | Ok |
Direct Dial In
Tested feature | Result |
---|---|
Inbound(Provider -> Innovaphone) | Ok |
Outbound(Innovaphone -> Provider) | Ok |
Loop In call(Innovaphone -> Provider -> Innovaphone) | Ok |
DTMF
Tested feature | Result |
---|---|
DTMF tones sent correctly via RTP-events(RFC 2833) | Ok |
DTMF tones sent correctly via SIP-Info | Nok |
DTMF tones received correctly via RTP-events(RFC 2833) | Ok |
Hold/Retrieve
Tested feature | Result |
---|---|
Call can be put on hold | Ok |
Held end hears music on hold / announcement from PBX | Ok |
Transfer with consultation
Tested feature | Result |
---|---|
Call can be transferred | Ok |
Held end hears music on hold | Ok |
The following tests are made to test if call transfer is working.
Tested feature | Voice Ok? | MoH Ok? |
---|---|---|
inno1 calls inno2. inno2 transfers to PSTN-phone. | Ok | Ok |
inno1 calls PSTN-phone. inno1 transfers to inno2. | Ok | Ok |
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. | Ok | Ok |
PSTN-phone calls inno1. inno1 transfers to inno2. | Ok | Ok |
PSTN-phone calls inno1. PSTN-phone transfers to inno2. | Ok | Ok |
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. | Ok | Ok |
Transfer with consultation (alerting only)
Tested feature | Result |
---|---|
Call can be transferred | Ok |
Held end hears music on hold or dialling tone | Ok |
Call returns to transferring device if the third
Endpoint is not available |
Ok |
The following tests are made to test if call transfer is working.
Tested feature | Voice Ok? | MoH Ok? |
---|---|---|
inno1 calls inno2. inno2 transfers to PSTN-phone. | Ok | Ok |
inno1 calls PSTN-phone. inno1 transfers to inno2. | Ok | Ok |
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. | Ok | Ok |
PSTN-phone calls inno1. inno1 transfers to inno2. | Ok | Ok |
PSTN-phone calls inno1. PSTN-phone transfers to inno2. | Ok | Ok |
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. | Ok | Ok |
Blind Transfer
Tested feature | Result |
---|---|
Call can be transferred | Ok |
Held end hears dialling tone | Ok |
The following tests are made to test if call transfer is working.
Tested feature | Voice Ok? |
---|---|
inno1 calls inno2. inno2 transfers to PSTN-phone. | Ok |
inno1 calls PSTN-phone. inno1 transfers to inno2. | Ok |
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. | Ok |
PSTN-phone calls inno1. inno1 transfers to inno2. | Ok |
PSTN-phone calls inno1. PSTN-phone transfers to inno2. | Ok |
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. | Ok |
CFU / CFB Transfer
Tested feature | Result |
---|---|
Call can be forwarded | Ok |
Held end hears dialling tone | Ok |
CFNR / Blind Transfer (alerting only)
Tested feature | Result |
---|---|
Call can be transferred or forwarded | Ok |
Held end hears dialling tone | Ok |
The following tests are made to test if call transfer is working.
Tested feature | Voice Ok? |
---|---|
inno1 calls inno2. inno2 transfers to PSTN-phone. | Ok |
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. | Ok |
PSTN-phone calls inno1. inno1 transfers to inno2. | Ok |
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. | Ok |
Broadcast Group & Waiting Queue
Tested feature | Result |
---|---|
Caller can make a call to a Broadcast Group | Ok |
Caller can make a call to a Waiting Queue | Ok |
Announcement if nobody picks up the call | Ok |
Configuration
Firmware version
All innovaphone devices use V11r2 SR1 as firmware.
SIP - Trunk
Number Mapping
Route Settings
Fax
Innovaphone Compatibility Test Report-old-test
Summary
SIP Provider: HFO
The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.
HFO 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.
HFO has achieved 89% of all possible test points. For more information on the test rating, please refer to Test Description
- Features:
- Direct Dial In
- DTMF
- Supported Codecs by the provider
- G711
- G729
Current test state
The tests for this product have been completed.
Testing of this product has been finalized October 22th, 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
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 | Yes |
Overlapped sending | No |
early media channel | Yes |
Fax using T.38 | No |
CGPN can be supressed | Yes |
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 |
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 |
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 | 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: HFO
- described in Mantis Case: 16505
Firmware version
- IP800: 6.00 dvl-sr2 IP800[07-60600.58]
- 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 HFO - 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.
HFO 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 HFO, 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.
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.