Howto:QSC IPfonie extended - SIP Provider Compatibility Test: Difference between revisions

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For more information on the test procedure, please read the following wiki article: [[Howto:SIP_Interop_Test_Description|SIP Interop Test Description]]. Bold lines in the test results indicate a KO-criteria.
For more information on the test procedure, please read the following wiki article: [[Howto:SIP_Interop_Test_Description|SIP Interop Test Description]]. Bold lines in the test results indicate a KO-criteria.


=== Basic Call ===
=== Basic Call ===
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|----
|----
|'''call using g711a'''
|'''call using g711a'''
|'''Yes'''
|
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|'''Yes'''
|
|----
|----
|call using g723
|call using g723
|Yes
|
|----
|----
|call using g729
|call using g729
|Yes
|
|----
|call using g722
|
|----
|----
|Overlapped sending
|Overlapped sending
|No
|
|----
|----
|'''early media channel'''
|'''early media channel'''
|'''Yes'''
|
|----
|----
|Fax using T.38
|Fax using T.38
|Yes
|
|----
|Reverse Media Negotiation
|
|----
|----
|CGPN can be supressed
|CGPN can be suppressed
|Yes
|
|----
|----
|CLIP no screening
|CLIP no screening
|Yes
|
|----
|'''Long time call possible(>30 min)'''
|
|----
|'''External Transfer'''
|
|----
|----
|'''Reverse Media Negotiaton'''
|NAT Detection
|'''Yes'''
|
|----
|----
|'''Long time call possible'''
|Redundancy
|'''Yes'''
|
|----
|----
|'''External Transfer'''
|SIP over TCP
|'''Yes'''
|
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|'''Yes'''
|
|}
|}


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|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|
|}
|}


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|----
|----
|'''DTMF tones sent correctly'''
|'''DTMF tones sent correctly'''
|'''Yes'''
|
|----
|DTMF tones sent correctly via SIP-Info
|
|----
|----
|'''DTMF tones received correctly'''
|'''DTMF tones received correctly'''
|'''Yes'''
|
|}
|}


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|----
|----
|'''Call can be put on hold'''  
|'''Call can be put on hold'''  
|'''Yes'''
|
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|Yes
|
|}
|}


Line 136: Line 150:
!Result  
!Result  
|----
|----
|'''Call can be transfered'''
|'''Call can be transferred'''
|'''Yes'''
|
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|Yes
|
|}
 
The following tests are made to test if call transfer is working.
 
{| border="1"
!Tested feature
!Voice Ok?
!MoH Ok?
|----
|----
|'''Call returns to transferring device if the third'''
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
'''Endpoint is not available'''
|
|'''Yes'''
|
|----
||inno1 calls inno2. inno1 transfers to sip-provider-phone.
|
|
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|
|}
|}


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!Result  
!Result  
|----
|----
|'''Call can be transfered'''
|'''Call can be transferred'''
|'''Yes'''
|
|----
|----
|Held end hears music on hold or dialing tone
|Held end hears music on hold or dialling tone
|Yes
|
|----
|----
|'''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'''
|
|}
 
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 sip-provider-phone.
|
|
|----
||inno1 calls inno2. inno1 transfers to sip-provider-phone.
|
|
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|
|}
|}


Line 170: Line 244:
!Result  
!Result  
|----
|----
|Call can be transfered
|Call can be transferred
|Yes
|
|----
|Held end hears dialling tone
|
|}
 
The following tests are made to test if call transfer is working.
 
{| border="1"
!Tested feature
!Voice Ok?
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|----
||inno1 calls inno2. inno1 transfers to sip-provider-phone.
|
|----
|inno1 calls sip-provider-phone. inno1 transfers to inno2.
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|----
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|
|----
|sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.
|
|}
 
=== Blind Transfer (alerting only)===
 
{| border="1"
!Tested feature
!Result
|----
|Call can be transferred
|
|----
|Held end hears dialling tone
|
|}
 
The following tests are made to test if call transfer is working.
 
{| border="1"
!Tested feature
!Voice Ok?
|----
|inno1 calls inno2. inno2 transfers to sip-provider-phone.
|
|----
|inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
|
|----
|----
|Held end hears dialing tone
|sip-provider-phone calls inno1. inno1 transfers to inno2.
|Yes
|
|}
|}


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|----
|----
|'''Caller can make a call to a Broadcast Group'''
|'''Caller can make a call to a Broadcast Group'''
|'''Yes'''
|
|----
|----
|'''Caller can make a call to a Waiting Queue'''
|'''Caller can make a call to a Waiting Queue'''
|'''Yes'''
|
|----
|----
|'''Announcement if nobody picks up the call'''
|'''Announcement if nobody picks up the call'''
|'''Yes'''
|
|}
|}


Revision as of 12:56, 4 October 2012

Innovaphone Compatibility Test Report

Summary

SIP Provider: QSC

  • Features:
    • Direct Dial In
    • DTMF
  • Supported Codecs by the provider
    • G711
    • G729
    • G723
    • T.38

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

The configuration on innovaphone side is simple, since the built-in SIP-GW can be used to connect to the provider.

QSC has achieved 98% of all possible test points. For more information on the test rating, please refer to Test Description

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 March 13th, 2010.

Testing Enviroment

Scenario NAT

HFO SIP Compatibility Test 5.PNG

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
call using g711u
call using g723
call using g729
call using g722
Overlapped sending
early media channel
Fax using T.38
Reverse Media Negotiation
CGPN can be suppressed
CLIP no screening
Long time call possible(>30 min)
External Transfer
NAT Detection
Redundancy
SIP over TCP
Voice Quality OK?

Direct Dial In

Tested feature Result
Inbound(Provider -> Innovaphone)
Outbound(Innovaphone -> Provider)

DTMF

Tested feature Result
DTMF tones sent correctly
DTMF tones sent correctly via SIP-Info
DTMF tones received correctly

Hold/Retrieve

Tested feature Result
Call can be put on hold
Held end hears music on hold / announcement from PBX

Transfer with consultation

Tested feature Result
Call can be transferred
Held end hears music on hold

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone.
inno1 calls inno2. inno1 transfers to sip-provider-phone.
inno1 calls sip-provider-phone. inno1 transfers to inno2.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.

Transfer with consultation (alerting only)

Tested feature Result
Call can be transferred
Held end hears music on hold or dialling tone
Call returns to transferring device if the third

Endpoint is not available

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone.
inno1 calls inno2. inno1 transfers to sip-provider-phone.
inno1 calls sip-provider-phone. inno1 transfers to inno2.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.

Blind Transfer

Tested feature Result
Call can be transferred
Held end hears dialling tone

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone.
inno1 calls inno2. inno1 transfers to sip-provider-phone.
inno1 calls sip-provider-phone. inno1 transfers to inno2.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.
sip-provider-phone calls inno1. sip-provider-phone transfers to inno2.

Blind Transfer (alerting only)

Tested feature Result
Call can be transferred
Held end hears dialling tone

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to sip-provider-phone.
inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2.
sip-provider-phone calls inno1. inno1 transfers to inno2.

Broadcast Group & Waiting Queue

Tested feature Result
Caller can make a call to a Broadcast Group
Caller can make a call to a Waiting Queue
Announcement if nobody picks up the call

Configuration

Firmware version

  • IP800: 8.00 dvl IP800[09-80394]
  • IP302: 8.00 dvl IP800[09-80394]
  • IP200: 8.00 dvl IP800[09-80394]
  • IP230: 8.00 dvl IP800[09-80394]

SIP - Trunk

First of all the SIP Trunk must be configured. Here an example of our QSC - 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. QSC 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.


QSC SIP Compatibility Test 1.PNG

Number Mapping

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

QSC SIP Compatibility Test 2.PNG

Route Settings

Because QSC, 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.

QSC SIP Compatibility Test 3.PNG

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. The tests described in this article, were succesfull without using Media Relay. If you run into problems(media, one way audio), you can still activate the Media Relay checkbox at the SIP interface.

Fax

The FAX was connected via a IP302 to the IP800. When configuring the IP302 you must keep in mind that you will use the analog interface for fax communication. Thats why the T.38 codec must be enabled.

QSC SIP Compatibility Test 5.PNG

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.

CLIR

To suppress the CGPN for outbound calls a config line option must be activated. Please enter the following lines in your browser: 

http://PBX-IP-address/!config add SIP /pai
http://PBX-IP-address/!config write
http://PBX-IP-address/!config activate

Related Articles

Tk Anlagenliste.pdf‎

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