Howto:TWT S.p.a. SIP Provider Compatibility Test: Difference between revisions
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The provider has achieved | 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 [[Howto:SIP_Interop_Test_Description#|Test Description]] | ||
<!-- Mention all important tests that were not passed in the summary. | <!-- Mention all important tests that were not passed in the summary. | ||
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** Fax over IP (T.38) | ** Fax over IP (T.38) | ||
** DTMF | ** DTMF | ||
** Reverse Media Negotiation | |||
* Supported Codecs by the provider | * Supported Codecs by the provider | ||
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This scenario describes a setup where the PBX and phones are in a private network. | 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 | |||
* the SIP trunk is configured with Media Relay | |||
== Test Results == | == Test Results == | ||
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|---- | |---- | ||
|Reverse Media Negotiation | |Reverse Media Negotiation | ||
| | |OK | ||
|---- | |---- | ||
|CGPN can be suppressed | |CGPN can be suppressed | ||
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[[Image:TWT_routing.png]] | [[Image:TWT_routing.png]] | ||
=== 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 [[Reference11r1:DHCP_client#Supported_Options | Default coder]]. The codec settings of interfaces must be configured manually. | |||
[[Category:Compat|{{PAGENAME}}]] | [[Category:Compat|{{PAGENAME}}]] |
Latest revision as of 07:57, 12 August 2015
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.