Howto:Swisscom Enterprise SIP - SIP Trunk - SIP Provider Compatibility Test: Difference between revisions

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TEST
'''Innovaphone Compatibility Test Report'''
 
== Summary ==
 
'''SIP Provider: swisscom Business Connect'''
 
The provider supports all mandatory innovaphone features and is therefore qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|recommended SIP Provider]].
 
This tests are not made by innovaphone, but by an external provider (tekvizion) with innovaphone support.
 
The Test results are documented by tekvizion
 
[[Documentation:Example.ogg]]
 
== Current test state ==
 
<!-- {{Template:Compat Status "in progress"}}-->
<!-- {{Template:Compat Status "planned"}} -->
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"engineered_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"rec._prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
 
{{Template:Compat Status "certified"|certificate=Swisscom_Business_Connect_-_SIP_Provider_-_product-cert.pdf}}
 
Testing of this product has been finalized February 2015.
 
== Testing Enviroment ==
 
[[Image:swisscomSIPProviderTestTopology1.png]]
 
This scenario describes a setup where the PBX and phones are in a private network.
 
The swisscom is providing an own Gateway ("Connect Box" or "Patton Smart Node") which must be located in the customers private network. This Gateway has two connections:
* One connection to swisscom
* One connection to the PBX
Because of this, the SIP Trunk does not connect directly to the internet, but to the swisscom Gateway ("Connect Box" or "Patton Smart Node").
 
The SIP trunk must be configured with "Media Relay", "exclusive G.711a" and "NO T.38".
 
[[Howto:Swisscom_-_SIP_Trunk_-_SIP_Provider_Compatibility_Test#Configuration|See configuration]]
 
== Test Results ==
 
For more information on the test procedure, please read the following wiki article: [[Howto:SIP_Interop_Test_Description_v11#|Test Description V11 and later]]. Bold lines in the test results indicate a KO-criteria.
 
(*1) To use Incoming Early Media Channel, this feature must be requested to swisscom. Per default, it's not possible.
 
(*2) Fax via G.711 (T.38 to G.711 / Audio Fax support) is not supported by swisscom but should work in most scenarios. This Feature is not guaranteed!
 
(*3) CLIP no screening does not work for "Trunk loop Connections". This means, if you dial out to the PSTN and back to your own Trunk Number, CLNS does not work in this case.
 
=== Basic Call ===
 
{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''OK'''
|----
|'''call using g711u'''
|'''NOK'''
|----
|call using g723
|NA
|----
|call using g729
|NA
|----
|call using g722
|NA
|----
|Overlapped sending
|NA
|----
|'''early media channel outgoing'''
|'''OK'''
|----
|early media channel incoming
|OK/NOK (*1)
|----
|Fax using T.38
|NA
|----
|T.38 Transcoding by the provider
|NA
|----
|Fax using G.711
|OK (*2)
|----
|Reverse Media Negotiation
|OK
|----
|CGPN can be suppressed
|OK
|----
|CLIP no screening
|OK (*3)
|----
|'''Long time call possible(>30 min)'''
|'''OK'''
|----
|'''External Transfer'''
|'''OK'''
|----
|NAT Detection
|NA
|----
|Redundancy
|NA
|----
|SIP over TCP
|NA
|----
|'''Voice Quality OK?'''
|'''OK'''
|}
 
=== Direct Dial In ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''OK'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''OK'''
|----
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|'''OK (*3)'''
|}
 
=== DTMF ===
 
{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly via RTP-events(RFC 2833)'''
|'''OK'''
|----
|DTMF tones sent correctly via SIP-Info
|OK
|----
|'''DTMF tones received correctly via RTP-events(RFC 2833)'''
|'''OK'''
|}
 
=== Hold/Retrieve ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''OK'''
|----
|Held end hears music on hold / announcement from PBX
|OK
|}
 
=== Transfer with consultation ===
 
{| border="1"
!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.
 
{| border="1"
!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) ===
 
{| border="1"
!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.
 
{| border="1"
!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 ===
 
{| border="1"
!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.
 
{| border="1"
!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 ===
 
{| border="1"
!Tested feature
!Result
|----
|'''Call can be forward'''
|'''OK'''
|----
|'''Held end hears dialling tone'''
|'''OK'''
|}
 
=== CFNR / Blind Transfer (alerting only)===
 
{| border="1"
!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.
 
{| border="1"
!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 ===
 
{| border="1"
!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 V11 build 11.0828 as firmware during test.
 
=== SIP - Trunk ===
Following configuration needs to be done in the SIP Gateway Interface settings:
 
* 1: Enter your phonenumber and private IP address of swisscom Gateway ("Connect Box" or "Patton Smart Node") with port 5062!
* 2: Enter credendtials
* 3: Select G.711a exclusive
* 4: Disable T.38 and Enable Media Relay
* 5: Disable encryption
* 6: Use "CGPN in user part of URI" in "From Header" and "Fixed AOR" in "Identity Header"
 
[[Image:swisscomSIPProviderTestConfig1.png]]
 
=== Number Mapping ===
Following configuration needs to be done in the SIP Gateway Number Mappings:
 
* 1: Add "00" to incoming CGPNs without a leading "0" because international numbers are coming in like "49703173009".
* 2: Because swisscom is using the full national number for incoming calls, you need to cut off the digits before an extension. In example, 04333693xx will be translated to xx.
* 3: In case of call forwardings, you need to cut off the single leading "0" (national calls) and translate a leading "000" (international calls) to "+" in the CGPN Out map.
* 4: In the CGPN Out map, you also need to translate the extensions to the national number format. In example, xx will be translated to 04333693xx.
 
[[Image:swisscomSIPProviderTestConfig2.png]]
 
=== Route Settings ===
A default Route Setup is used:
 
* 1: Force Enblock Setting is required for outgoing calls.
 
[[Image:swisscomSIPProviderTestConfig3.png]]
 
=== Fax ===
'''swisscom does not support any Fax communications.''' Because of this, T.38 can not be used.
 
If you want to use Fax even so, you need to set T.38 to G.711 translation (Audio Fax support, available from V11). This works in a lot of cases where you have no problems with QoS. '''But it's not guaranteed and swisscom gives no support in case of problems!''' 
 
Example to set T.38 to G.711 translation (Audio Fax support) on the Faxserver interface:
 
[[Image:swisscomSIPProviderTestConfig4.png]]
 
[[Category:Compat|{{PAGENAME}}]]

Revision as of 12:09, 9 August 2016

Innovaphone Compatibility Test Report

Summary

SIP Provider: swisscom Business Connect

The provider supports all mandatory innovaphone features and is therefore qualified as recommended SIP Provider.

This tests are not made by innovaphone, but by an external provider (tekvizion) with innovaphone support.

The Test results are documented by tekvizion

Documentation:Example.ogg

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 February 2015.

Testing Enviroment

SwisscomSIPProviderTestTopology1.png

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

The swisscom is providing an own Gateway ("Connect Box" or "Patton Smart Node") which must be located in the customers private network. This Gateway has two connections:

  • One connection to swisscom
  • One connection to the PBX

Because of this, the SIP Trunk does not connect directly to the internet, but to the swisscom Gateway ("Connect Box" or "Patton Smart Node").

The SIP trunk must be configured with "Media Relay", "exclusive G.711a" and "NO T.38".

See configuration

Test Results

For more information on the test procedure, please read the following wiki article: Test Description V11 and later. Bold lines in the test results indicate a KO-criteria.

(*1) To use Incoming Early Media Channel, this feature must be requested to swisscom. Per default, it's not possible.

(*2) Fax via G.711 (T.38 to G.711 / Audio Fax support) is not supported by swisscom but should work in most scenarios. This Feature is not guaranteed!

(*3) CLIP no screening does not work for "Trunk loop Connections". This means, if you dial out to the PSTN and back to your own Trunk Number, CLNS does not work in this case.

Basic Call

Tested feature Result
call using g711a OK
call using g711u NOK
call using g723 NA
call using g729 NA
call using g722 NA
Overlapped sending NA
early media channel outgoing OK
early media channel incoming OK/NOK (*1)
Fax using T.38 NA
T.38 Transcoding by the provider NA
Fax using G.711 OK (*2)
Reverse Media Negotiation OK
CGPN can be suppressed OK
CLIP no screening OK (*3)
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection NA
Redundancy NA
SIP over TCP NA
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 (*3)

DTMF

Tested feature Result
DTMF tones sent correctly via RTP-events(RFC 2833) OK
DTMF tones sent correctly via SIP-Info OK
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 V11 build 11.0828 as firmware during test.

SIP - Trunk

Following configuration needs to be done in the SIP Gateway Interface settings:

  • 1: Enter your phonenumber and private IP address of swisscom Gateway ("Connect Box" or "Patton Smart Node") with port 5062!
  • 2: Enter credendtials
  • 3: Select G.711a exclusive
  • 4: Disable T.38 and Enable Media Relay
  • 5: Disable encryption
  • 6: Use "CGPN in user part of URI" in "From Header" and "Fixed AOR" in "Identity Header"

SwisscomSIPProviderTestConfig1.png

Number Mapping

Following configuration needs to be done in the SIP Gateway Number Mappings:

  • 1: Add "00" to incoming CGPNs without a leading "0" because international numbers are coming in like "49703173009".
  • 2: Because swisscom is using the full national number for incoming calls, you need to cut off the digits before an extension. In example, 04333693xx will be translated to xx.
  • 3: In case of call forwardings, you need to cut off the single leading "0" (national calls) and translate a leading "000" (international calls) to "+" in the CGPN Out map.
  • 4: In the CGPN Out map, you also need to translate the extensions to the national number format. In example, xx will be translated to 04333693xx.

SwisscomSIPProviderTestConfig2.png

Route Settings

A default Route Setup is used:

  • 1: Force Enblock Setting is required for outgoing calls.

SwisscomSIPProviderTestConfig3.png

Fax

swisscom does not support any Fax communications. Because of this, T.38 can not be used.

If you want to use Fax even so, you need to set T.38 to G.711 translation (Audio Fax support, available from V11). This works in a lot of cases where you have no problems with QoS. But it's not guaranteed and swisscom gives no support in case of problems!

Example to set T.38 to G.711 translation (Audio Fax support) on the Faxserver interface:

SwisscomSIPProviderTestConfig4.png

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