Howto:HFO NGN Connect - HFO - SIP Provider: Difference between revisions

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The provider supports all required innovaphone features and is therefore qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|recommended SIP Provider]].  
The provider supports all required innovaphone features and is therefore qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|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.  
Only if the Mobility-feature or WebRTC is desired by the customer, you must enable Media-Relay on the SIP-Trunk. 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.


HFO has achieved 89% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]]
The provider doesn't support the G.729 coder, this might be an issue in locations with low-bandwidth Internet connection.
 
The PBX is connected to the provider through the public Internet, the tested product "HFO NGN Connect" comes without a dedicated Internet Access Device.
According to the provider the amount of concurrent calls is limited when ordering the trunk, currently max. 220 concurrent calls are possible.
 
The provider offers an additional SIP-trunk product called [http://www.hfo-telecom.de/produkte/geschaeftskunden/hfo-ngn-business HFO NGN Business]. This product comes with a dedicated Internet Access, used only for telephony. According to HFO, it uses the same SIP server as the tested product "HFO NGN Connect". As a result, the configuration on innovaphone side should be equal. For more details on the product "HFO NGN Business", please contact the provider directly.
 
NAT - Detection doesn't work for all private networks, see [[Howto:HFO_NGN_Connect_-_HFO_-_SIP_Provider#Known_Problems | Known Problems]] for more information.
 
That being said, the provider has achieved 92% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#|Test Description]]


* Features:
* Features:


** Direct Dial In
** Direct Dial In
** Fax over IP (T.38)
** DTMF
** DTMF
** Clip No Screening


* Supported Codecs by the provider
* Supported Codecs by the provider
** G711
** G711
** G729
** G722
** T.38 UDP


== Current test state ==
== Current test state ==
{{Template:Compat Status "tested"}}
 
<!--{{Template:Compat Status "planned"}} -->
<!-- {{Template:Compat Status "in progress"}} -->
<!-- {{Template:Compat Status "in progress"}} -->
<!--{{Template:Compat Status "certified"|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}}-->
<!-- {{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=HFO_NGN_Connect_HFO_SIP_Provider_-_product-cert.pdf}}
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
Testing of this product has been finalized June 23th, 2015.


Testing of this product has been finalized October 22th, 2007.
== Testing Environment ==


== Testing Enviroment ==
[[Image:SIPProviderTestTopology1.PNG]]


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


=== Scenario NAT ===


[[Image:HFO_SIP_Compatibility_Test_5.PNG]]
* the SIP trunk is configured without Media Relay and without exclusive coder. Media-Relay must be only activated if ''Mobility'' is used.


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 ==
== Test Results ==
Line 46: Line 64:
!Tested feature
!Tested feature
!Result  
!Result  
|----
|SIP over TLS(SIPS)
|Nok
|----
|SIP over TCP
|Nok
|----
|SRTP
|Nok
|----
|----
|'''call using g711a'''
|'''call using g711a'''
|'''Yes'''
|'''Ok'''
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|'''Yes'''
|'''Ok'''
|----
|----
|call using g723
|call using g729
|No
|Nok
|----
|----
|call using g729
|call using g722
|Yes
|Ok
|----
|----
|Overlapped sending
|Overlapped sending
|No
|Nok
|----
|----
|'''early media channel'''
|'''early media channel'''
|'''Yes'''
|'''Ok'''
|----
|----
|Fax using T.38
|Fax using T.38
|No
|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'''
|----
|----
|CGPN can be supressed
|'''External Transfer'''
|Yes
|'''Ok'''
|----
|----
|'''Reverse Media Negotiaton'''
|NAT Detection
|'''Yes'''
|Ok*
|----
|Redundancy
|Ok
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|'''Yes'''
|'''Ok'''
|}
|}
* NAT - Detection doesn't work for all private networks, see section [[Howto:HFO_NGN_Connect_-_HFO_-_SIP_Provider#Known_Problems | Known Problems]] for more information.


=== Direct Dial In ===
=== Direct Dial In ===
Line 85: Line 132:
|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|'''Ok'''
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|'''Ok'''
|----
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|'''Ok'''
|}
|}


Line 97: Line 147:
!Result  
!Result  
|----
|----
|'''DTMF tones sent correctly'''
|'''DTMF tones sent correctly via RTP-events(RFC 2833)'''
|'''Yes'''
|'''Ok'''
|----
|----
|'''DTMF tones received correctly'''
|DTMF tones sent correctly via SIP-Info
|'''Yes'''
|Nok
|----
|'''DTMF tones received correctly via RTP-events(RFC 2833)'''
|'''Ok'''
|}
|}


Line 111: Line 164:
|----
|----
|'''Call can be put on hold'''  
|'''Call can be put on hold'''  
|'''Yes'''
|'''Ok'''
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|Yes
|Ok
|----
|Held end hears music on hold / announcement from provider
|No
|}
|}


Line 126: Line 176:
!Result  
!Result  
|----
|----
|'''Call can be transfered'''
|'''Call can be transferred'''
|'''Yes'''
|'''Ok'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|Yes
|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
|----
|----
|'''Call returns to transferring device if the third'''
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
'''Endpoint is not available'''
|Ok
|'''Yes'''
|Ok
|}
|}


Line 143: Line 221:
!Result  
!Result  
|----
|----
|'''Call can be transfered'''
|'''Call can be transferred'''
|'''Yes'''
|'''Ok'''
|----
|----
|Held end hears music on hold or dialing tone
|Held end hears music on hold or dialling tone
|Yes
|Ok
|----
|----
|'''Call returns to transferring device if the third'''  
|'''Call returns to transferring device if the third'''  
'''Endpoint is not available'''
'''Endpoint is not available'''
|'''Yes'''
|'''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
|}
|}


Line 160: Line 270:
!Result  
!Result  
|----
|----
|Call can be transfered
|Call can be transferred
|Yes
|Ok
|----
|----
|Held end hears dialing tone
|Held end hears dialling tone
|No
|Ok
|}
|}


=== Broadcast Group & Waiting Queue ===
The following tests are made to test if call transfer is working.


{| border="1"  
{| border="1"  
!Tested feature
!Tested feature
!Result
!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
|----
|----
|'''Caller can make a call to a Broadcast Group'''
|PSTN-phone calls inno1. inno1 transfers to inno2.
|'''Yes'''
|Ok
|----
|----
|'''Caller can make a call to a Waiting Queue'''
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|'''Yes'''
|Ok
|----
|----
|'''Announcement if nobody picks up the call'''
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|'''Yes'''
|Ok
|}
|}


== Configuration ==
=== CFU / CFB Transfer ===


=== General Information ===
{| border="1"
!Tested feature
!Result
|----
|'''Call can be forwarded'''
|Ok
|----
|'''Held end hears dialling tone'''
|Ok
|}


'''Basic Provider Infomation: HFO'''
=== CFNR / Blind Transfer (alerting only)===


{| border="1"
!Tested feature
!Result
|----
|'''Call can be transferred or forwarded'''
|'''Ok'''
|----
|'''Held end hears dialling tone'''
|'''Ok'''
|}


*described in Mantis Case: 16505
The following tests are made to test if call transfer is working.


'''Firmware version'''
{| 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 ===


*IP800: 6.00 dvl-sr2 IP800[07-60600.58]
{| border="1"
*IP22: 6.00 dvl-sr1 IP230[07-60600.58]
!Tested feature
*IP200: 6.00 dvl-sr1 IP230[07-60600.58]
!Result
*IP230: 6.00 dvl-sr1 IP230[07-60600.58]
|----
|'''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'''
|}


=== SIP - Trunk ===
== Configuration ==


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.
===Firmware version===


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''.
All innovaphone devices use V11r2 SR1 as firmware.


=== SIP - Trunk ===
First of all the SIP Trunk must be configured. Here an example of our HFO - Trunk. If the Mobility-feature is desired by the customer, you must enable ''Media-Relay'' on the SIP-Trunk.


[[Image:HFO SIP Compatibility Test 1.PNG]]
[[Image:HFO SIP.PNG]]


=== Number Mapping ===
=== Number Mapping ===
The provider uses a CDPN in national format for incoming call. Therefore the SIP-interface maps must be configured accordingly.
[[Image:HFO SIP Compatibility Test 2.PNG]]


The complicated part on this issue is the correct mapping of the outgoing and incoming numbers.


[[Image:HFO SIP Compatibility Test 2.PNG]]


=== Route Settings ===
=== 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.
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.


[[Image:HFO SIP Compatibility Test 3.PNG]]
[[Image:HFO SIP Compatibility Test 3.PNG]]


=== Media Relay ===
If there is a problem with the number format, make sure to read also [[Howto:Create_ClipNoScreening_Maps_for_SIP_Interfaces]]
 
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.
 
[[Image:HFO SIP Compatibility Test 4.PNG]]
 
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.
=== Fax ===
The provider supports T.38, make sure to enable it on the SIP-interface and also on all analogue - interfaces used for fax-machines .


== Known Problems ==
HFO doesn't support NAT Traversal for clients using private addresses in 172.16.0.0/24 to 172.31.0.0/24 networks. In case that this private networks are used, STUN must be used at the RTP endpoints.  In order for NAT-Traversal using STUN to work, the [[Reference11r1:Concept_Using_PBX_services_from_public_internet#Detecting_the_NAT_style |NAT-Router must support full-cone or restricted NAT]].


[[Category:Compat|{{PAGENAME}}]]
[[Category:Compat|{{PAGENAME}}]]

Latest revision as of 14:40, 30 September 2020

Innovaphone Compatibility Test Report

Summary

SIP Provider: HFO

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

Only if the Mobility-feature or WebRTC is desired by the customer, you must enable Media-Relay on the SIP-Trunk. 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.

The provider doesn't support the G.729 coder, this might be an issue in locations with low-bandwidth Internet connection.

The PBX is connected to the provider through the public Internet, the tested product "HFO NGN Connect" comes without a dedicated Internet Access Device. According to the provider the amount of concurrent calls is limited when ordering the trunk, currently max. 220 concurrent calls are possible.

The provider offers an additional SIP-trunk product called HFO NGN Business. This product comes with a dedicated Internet Access, used only for telephony. According to HFO, it uses the same SIP server as the tested product "HFO NGN Connect". As a result, the configuration on innovaphone side should be equal. For more details on the product "HFO NGN Business", please contact the provider directly.

NAT - Detection doesn't work for all private networks, see Known Problems for more information.

That being said, the provider has achieved 92% of all possible test points. For more information on the test rating, please refer to Test Description

  • Features:
    • Direct Dial In
    • Fax over IP (T.38)
    • DTMF
    • Clip No Screening
  • Supported Codecs by the provider
    • G711
    • G722
    • T.38 UDP

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 June 23th, 2015.

Testing Environment

SIPProviderTestTopology1.PNG

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


  • the SIP trunk is configured without Media Relay and without exclusive coder. Media-Relay must be only activated if Mobility is used.


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 Ok*
Redundancy Ok
Voice Quality OK? Ok 
* NAT - Detection doesn't work for all private networks, see section  Known Problems for more information.

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

First of all the SIP Trunk must be configured. Here an example of our HFO - Trunk. If the Mobility-feature is desired by the customer, you must enable Media-Relay on the SIP-Trunk.

HFO SIP.PNG

Number Mapping

The provider uses a CDPN in national format for incoming call. Therefore the SIP-interface maps must be configured accordingly.

HFO SIP Compatibility Test 2.PNG


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.

HFO SIP Compatibility Test 3.PNG

If there is a problem with the number format, make sure to read also Howto:Create_ClipNoScreening_Maps_for_SIP_Interfaces

Fax

The provider supports T.38, make sure to enable it on the SIP-interface and also on all analogue - interfaces used for fax-machines .

Known Problems

HFO doesn't support NAT Traversal for clients using private addresses in 172.16.0.0/24 to 172.31.0.0/24 networks. In case that this private networks are used, STUN must be used at the RTP endpoints. In order for NAT-Traversal using STUN to work, the NAT-Router must support full-cone or restricted NAT.

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