Howto:TWT S.p.a. SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
Summary
SIP Provider: TWT S.p.a.
The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.
The provider has achieved 84% (135 out of 161 points) of all possible test points. For more information on the test rating, please refer to Test Description
This provider supports DDI only through a trunk configuration without authentication. For this reason a Gateway interface w/out authentication must be used. Also NAT detection is not supported so media relay with exclusive coder and a STUN server must be configured (port forwarding for port 5060 must be configured on NAT Router toward local gateway ip address).
Additionally phones must be configured to use a coder framesize of 20 ms since this is the only value supported by TWT.
Since media relay and an exclusive coder setting must be configured, opposed to a SIP trunk not needing Media-Relay, the transport of all RTP packets by the gateway will result in a higher CPU load for a call. As a result, the amount of concurrent calls is considerably lower compared to a SIP-Provider that doesn't require Media-Relay.
- Features:
- Direct Dial In
- Fax over IP (T.38)
- DTMF
- Reverse Media Negotiation
- Supported Codecs by the provider
- G711a
- G711u
- G729
- G722
- T.38 UDP
Current test state
This product is listed due to a customer testimonial. No tests have been conducted by innovaphone.
Testing of this product has been finalized May 12, 2015.
Testing Enviroment
This scenario describes a setup where the PBX and phones are in a private network.
- the SIP trunk is configured with Media Relay and STUN . This is the case when the test for "NAT Traversal" fails
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) | N/A |
| SIP over TCP | N/A |
| SRTP | NOK |
| call using g711a | OK |
| call using g711u | OK |
| call using g723 | NOK |
| call using g729 | OK |
| call using g722 | OK |
| Overlapped sending | NOK |
| early media channel | OK |
| Fax using T.38 | OK |
| T.38 Transcoding by the provider | NOK |
| Fax using G.711 | OK |
| Reverse Media Negotiation | OK |
| CGPN can be suppressed | NOK |
| CLIP no screening | NOK |
| Long time call possible(>30 min) | OK |
| External Transfer | OK |
| NAT Detection | NOK |
| Redundancy | OK* |
| Voice Quality OK? | OK |
* Note to Redundancy: the provider can send signalling to a secondary public ip address if primary does not respond.
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 | N/A |
| 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 forward | OK |
| Held end hears dialling tone | OK |
CFNR / Blind Transfer (alerting only)
| Tested feature | Result |
|---|---|
| Call can be transferred or forward | 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 build 11.3127 as firmware.
SIP - Trunk
Trunk Interface
Number Mapping
Route Settings
Codec/Framesize
TWT accepts only RTP-packets having a Framesize of 20ms. You must configure all RTP-endpoints(e.g. phones, analog adapters, ISDN interfaces, etc.) to use 20ms as Framesize. For phones you can use a DHCP-server to distribute the Default coder. The codec settings of interfaces must be configured manually.