Howto:GlobalConnect SIP-Trunk - SIP Provider Compatibility Test: Difference between revisions

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== Summary ==
== Summary ==


'''SIP Provider: Telavox'''
'''GlobalConnect'''
 
<!--


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]].  


The provider doesn't support [[Howto:SIP_Interop_Test_Description#Redundancy_Mechanism.28Important.29 | Redundancy]] scenarios with multiple SIP-trunks(master & standby) registered at one account.


...
The provider also doesn't support [[Howto:SIP_Interop_Test_Description#T.38_Transcoding_by_the_provider.28Important.29 | T.38 Transcoding]].


That being said, the provider has achieved x% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#|Test Description]]
That being said, the provider has achieved 96% of all possible test points (150/157). 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.


E.g. in case that the provider doesn't support Reverse Media-Negotiation, mention in the summary that media relay and an exclusive coder setting must be configured:
Since the provider doesn't support Reverse Media negotiation, 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.
-->
<!-- in case of customer testimonial, please make sure that the fact that it is not tested by innovaphone is also mentioned in the summary-->
<!--
* Features:
* Features:


** Direct Dial In
** Fax over IP (T.38)
** DTMF
** DTMF


Line 31: Line 20:
** G711
** G711
** G729
** G729
** G722
** G723
** G723
** G726
** T.38
** T.38 UDP -->


== Current test state ==
== Current test state ==


<!--{{Template:Compat Status "planned"}} -->
<!--{{Template:Compat Status "planned"}} -->
{{Template:Compat Status "in progress"}}
<!-- {{Template:Compat Status "in progress"}} -->
<!-- {{Template:Compat_Status_"referral_prod."|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_"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_"rec._prod."|certificate=SIP-Trunk_GlobalConnect_SIP_Provider_-_product-cert.pdf}}
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "tested"(sip provider)}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


<!-- Testing of this product has been finalized January 1st, 1970. -->
Testing of this product has been finalized September 2nd, 2014.


== Testing Enviroment ==
== Testing Enviroment ==
Line 54: Line 43:
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 without Media Relay and without exclusive coder. This is the case if all tests were successful
The SIP trunk is configured without Media Relay and without exclusive coder.


== Test Results ==
== Test Results ==
Line 67: Line 56:
|----
|----
|'''call using g711a'''
|'''call using g711a'''
|
|'''Ok'''
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|
|'''Ok'''
|----
|----
|call using g723
|call using g723
|
|Ok, if remote end is at same provider and also supports G723
|----
|----
|call using g729
|call using g729
|
|Ok
|----
|----
|call using g722
|call using g722
|
|Ok, if remote end is at same provider and also supports G722
|----
|----
|Overlapped sending
|Overlapped sending
|
|Nok
|----
|----
|'''early media channel'''
|'''early media channel'''
|
|'''Ok'''
|----
|----
|Fax using T.38
|Fax using T.38
|
|Ok
|----
|----
|T.38 Transcoding by the provider
|T.38 Transcoding by the provider
|
|Nok
|----
|----
|Reverse Media Negotiation
|Reverse Media Negotiation
|
|Ok
|----
|----
|CGPN can be suppressed
|CGPN can be suppressed
|
|Ok
|----
|----
|CLIP no screening
|CLIP no screening
|
|Ok
|----
|----
|'''Long time call possible(>30 min)'''
|'''Long time call possible(>30 min)'''
|
|'''Ok'''
|----
|----
|'''External Transfer'''
|'''External Transfer'''
|
|'''Ok'''
|----
|----
|NAT Detection
|NAT Detection
|
|Ok
|----
|----
|Redundancy
|Redundancy
|
|Nok
|----
|----
|SIP over TCP
|SIP over TCP
|
|Ok
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|
|'''Ok'''
|}
|}


Line 128: Line 117:
|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone)'''
|
|'''Ok'''
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|
|'''Ok'''
|----
|----
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|
|Ok
|}
|}


Line 143: Line 132:
!Result  
!Result  
|----
|----
|'''DTMF tones sent correctly via RTP-events(RFC 2833)'''
|'''DTMF tones sent correctly'''
|
|'''Ok'''
|----
|----
|DTMF tones sent correctly via SIP-Info
|DTMF tones sent correctly via SIP-Info
|
|Nok
|----
|----
|'''DTMF tones received correctly via RTP-events(RFC 2833)'''
|'''DTMF tones received correctly'''
|
|'''Ok'''
|}
|}


Line 160: Line 149:
|----
|----
|'''Call can be put on hold'''  
|'''Call can be put on hold'''  
|
|'''Ok'''
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|
|Ok
|}
|}


Line 173: Line 162:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''Ok'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|
|Ok
|}
|}


Line 187: Line 176:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|Ok
|
|Ok
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|Ok
|
|Ok
|}
|}


Line 218: Line 207:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''Ok'''
|----
|----
|Held end hears music on hold or dialling tone
|Held end hears music on hold or dialling tone
|
|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'''
|
|'''Ok'''
|}
|}


Line 236: Line 225:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|Ok
|
|Ok
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|Ok
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|Ok
|
|Ok
|}
|}


Line 267: Line 256:
|----
|----
|Call can be transferred
|Call can be transferred
|
|Ok
|----
|----
|Held end hears dialling tone
|Held end hears dialling tone
|
|Ok
|}
|}


Line 280: Line 269:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|Ok
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|Ok
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|Ok
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|Ok
|}
|}


Line 304: Line 293:
!Result  
!Result  
|----
|----
|'''Call can be forward'''
|'''Call can be forwarded'''
|
|'''Ok'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''Ok'''
|}
|}


Line 317: Line 306:
!Result  
!Result  
|----
|----
|'''Call can be transferred or forward'''
|'''Call can be transferred or forwarded'''
|
|'''Ok'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''Ok'''
|}
|}


Line 331: Line 320:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|Ok
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|Ok
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|Ok
|}
|}


Line 350: Line 339:
|----
|----
|'''Caller can make a call to a Broadcast Group'''
|'''Caller can make a call to a Broadcast Group'''
|
|'''Ok'''
|----
|----
|'''Caller can make a call to a Waiting Queue'''
|'''Caller can make a call to a Waiting Queue'''
|
|'''Ok'''
|----
|----
|'''Announcement if nobody picks up the call'''
|'''Announcement if nobody picks up the call'''
|
|'''Ok'''
|}
|}


Line 363: Line 352:
===Firmware version===
===Firmware version===


All innovaphone devices use Vx build xx-xxxxx as firmware.
All innovaphone devices use V10 SR12 as firmware.


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


[[Image:GlobalConnect_SIP-Trunk_-_SIP_Provider_Compatibility_Test_1.png]]


A SIP interface can be used to connect to the provider. Make sure to configure the ''SIP Interop Tweaks'' as in the screenshot above. MediaRelay is not needed.
=== Number Mapping ===
=== Number Mapping ===


[[Image:GlobalConnect_SIP-Trunk_-_SIP_Provider_Compatibility_Test_2.png]]
The provider will send and receive all numbers (CGPN & CDPN) in international number format.


=== Route Settings ===
=== Route Settings ===


[[Image:GlobalConnect_SIP-Trunk_-_SIP_Provider_Compatibility_Test_3.png]]
The provider does not support overlap dialling, so ''Force Enblock'' must be activated in the route from ''RS1'' to ''SIP1''.


=== Media Relay ===
Since ''Clip No Screening'' is supported, make sure to have a route mapping that adjust calls with numbers other than the own trunknumber to a proper format. (see screenshot comment Clip No Screening)


=== Known Problems ===


=== Fax ===
* In order for '''CLIP No Screening''' to work, we must configure the use of the ''P-Asserted Identity'' as shown below.


http://x.x.x.x/!config add SIP /pai
http://x.x.x.x/!config write
http://x.x.x.x/!config activate


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

Revision as of 16:07, 9 December 2014

Innovaphone Compatibility Test Report

Summary

GlobalConnect

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

The provider doesn't support Redundancy scenarios with multiple SIP-trunks(master & standby) registered at one account.

The provider also doesn't support T.38 Transcoding.

That being said, the provider has achieved 96% of all possible test points (150/157). For more information on the test rating, please refer to Test Description

  • Features:
    • DTMF
  • Supported Codecs by the provider
    • G711
    • G729
    • G722
    • G723
    • T.38

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 September 2nd, 2014.

Testing Enviroment

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.

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 Ok
call using g711u Ok
call using g723 Ok, if remote end is at same provider and also supports G723
call using g729 Ok
call using g722 Ok, if remote end is at same provider and also supports G722
Overlapped sending Nok
early media channel Ok
Fax using T.38 Ok
T.38 Transcoding by the provider Nok
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 Nok
SIP over TCP Ok
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

DTMF

Tested feature Result
DTMF tones sent correctly Ok
DTMF tones sent correctly via SIP-Info Nok
DTMF tones received correctly 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 V10 SR12 as firmware.

SIP - Trunk

GlobalConnect SIP-Trunk - SIP Provider Compatibility Test 1.png

A SIP interface can be used to connect to the provider. Make sure to configure the SIP Interop Tweaks as in the screenshot above. MediaRelay is not needed.

Number Mapping

GlobalConnect SIP-Trunk - SIP Provider Compatibility Test 2.png

The provider will send and receive all numbers (CGPN & CDPN) in international number format.

Route Settings

GlobalConnect SIP-Trunk - SIP Provider Compatibility Test 3.png

The provider does not support overlap dialling, so Force Enblock must be activated in the route from RS1 to SIP1.

Since Clip No Screening is supported, make sure to have a route mapping that adjust calls with numbers other than the own trunknumber to a proper format. (see screenshot comment Clip No Screening)

Known Problems

  • In order for CLIP No Screening to work, we must configure the use of the P-Asserted Identity as shown below.
http://x.x.x.x/!config add SIP /pai
http://x.x.x.x/!config write
http://x.x.x.x/!config activate
Retrieved from "https://wiki.innovaphone.com/index.php?title=Howto:GlobalConnect_SIP-Trunk_-_SIP_Provider_Compatibility_Test&oldid=36183"