Howto:Uni-tel SIP Provider Compatibility Test: Difference between revisions

From innovaphone wiki
Jump to navigation Jump to search
VisualWikitext
m New page: == Summary == '''SIP Provider: Deltacom''' The provider has achieved 79,17% of all possible test points (95 on total of 120 Points). For more information on the test rating, please refer...
 
 
(39 intermediate revisions by 2 users not shown)
Line 1: Line 1:
== Summary ==
== Summary ==


'''SIP Provider: Deltacom'''
'''SIP Provider: Uni-tel A/S'''


The provider has achieved 79,17% of all possible test points (95 on total of 120 Points). For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]]
The provider has achieved 93,24% of all possible test points (124 on total of 133 Points). For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]]




Line 16: Line 16:
* Supported Codecs by the provider
* Supported Codecs by the provider
** G711 a/u
** G711 a/u
** G729


== Current test state ==
== Current test state ==
<!-- {{Template:Compat Status "tested"}} -->
<!-- {{Template:Compat Status "tested"}} -->
<!-- {{Template:Compat Status "in progress"}}-->
<!-- {{Template:Compat Status "in progress"}} -->  
{{Template:Compat Status "certified"|certificate=load_certificate.pdf}}
{{Template:Compat Status "certified"|certificate=IP_Telefoni_Uni-Tel_SIP_Provider_-_product-cert.pdf}}
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat Status "rejected"}} -->


Testing of this product has been finalized in August, 2012.
Testing of this product has been finalized in 28th August, 2012.


== Testing Enviroment ==
== Testing Enviroment ==
Line 32: Line 31:
[[Image:HFO_SIP_Compatibility_Test_5.PNG]]
[[Image:HFO_SIP_Compatibility_Test_5.PNG]]


This scenario describes a setup where the PBX and phones are in a private network. The IP800 should use a stun server, in order to send correct SIP - messages. Media-relay on IP800 SIP Trunk Interface to provider is also required, all RTP-streams go through the PBX.  
This scenario describes a setup where the PBX and phones are in a private network.


== Test Results ==
== Test Results ==
Line 51: Line 50:
|----
|----
|Call using g723
|Call using g723
|X
|No
|----
|----
|Call using g729
|Call using g729
|Yes
|No
|----
|----
|Call using g722
|Call using g722
|X
|No
|----
|----
|Overlapped sending
|Overlapped sending
|X
|No
|----
|----
|'''Early media channel'''
|'''Early media channel'''
Line 66: Line 65:
|----
|----
|Fax using T.38
|Fax using T.38
|X
|Yes
|----
|----
|CGPN can be suppressed
|CGPN can be suppressed
|X
|Yes
|----
|----
|CLIP no screening  
|CLIP no screening  
|X
|Yes (only for Danish numbers)
|----
|----
|Reverse Media Negotiation
|Reverse Media Negotiation
|X(Media-relay necessary)
|Yes
|----
|----
|'''Long time call possible'''
|'''Long time call possible'''
Line 84: Line 83:
|----
|----
|NAT Detection
|NAT Detection
|X
|Yes
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
Line 90: Line 89:
|----
|----
|Redundancy Mechanism
|Redundancy Mechanism
|Yes
|----
|SIP over TCP
|Yes
|Yes
|}
|}
Line 204: Line 206:
Here's the configuration of the SIP gateway interface.
Here's the configuration of the SIP gateway interface.


[[Image:Deltacom_-_TRUNK_SETUP.png]]
[[Image:Uni-tel_SIP_Provider_Compatibility_Test_1.png]]


Deltacom 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.  
No need to use STUN, so It's optional.  
 
Fixed AOR should not be used to send different CPGN since only "FROM" header it's used for CGPN and not Prefered-ID field on SIP Invite.


=== Number Mapping ===
=== Number Mapping ===


[[Image:Deltacom_-_NUMBER_MAPPING.png]]
[[Image:Uni-tel_SIP_Provider_Compatibility_Test_2.png]]


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


Deltacom, as most SIP Providers do, doesn't support overlap sending. You must enable the enblock sending of the phone number. You can do this by enabling Force enblock in the appropriate outgoing Route.
Uni-tel, as most SIP Providers do, doesn't support overlap sending. You must enable the enblock sending of the phone number. You can do this by enabling '''Force enblock''' in the appropriate outgoing Route.


The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this check-box is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.  
The second setting you must check is '''Interworking(QSIG,SIP)'''. This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this check-box is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.  
 
[[Image:Deltacom_-_ROUTING.png]]


[[Image:Uni-tel_SIP_Provider_Compatibility_Test_3.png]]


=== Redundancy ===
=== Redundancy ===


Redundancy is supported by configuring a dedicated GW interface on STAND-BY device. Provider allows up to 3 concurrent interfaces configured for a single account. Incoming calls will be sent parallel on both MASTER and STAND-BY but will be answered only by the MASTER when up.
Redundancy could be done by DNS SRV.


When resolving the proxy name the provider may give us several IP address in the SRV record.
We select one address and keep the others as fallback.
Methods used to select one address of several returned addresses are also used for load balancing.
Whether load balancing and/or redundancy is implemented only depends on priority values of the returned addresses.


=== Known Issues ===
=== Known Issues ===
CGPN for outgoing calls cannot be suppressed as the provider only allows calls from the assigned number range.
We were not able to have T38 work correctly with this provider, however there were no problems sending and receiving faxes without it.

Latest revision as of 15:24, 31 August 2012

Summary

SIP Provider: Uni-tel A/S

The provider has achieved 93,24% of all possible test points (124 on total of 133 Points). For more information on the test rating, please refer to Test Description


  • Features:
    • Direct Dial In
    • DTMF
    • Redundancy Mechanism


  • Supported Codecs by the provider
    • G711 a/u

Current test state

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 in 28th August, 2012.

Testing Enviroment

Scenario NAT

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

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 Yes
Call using g711u Yes
Call using g723 No
Call using g729 No
Call using g722 No
Overlapped sending No
Early media channel Yes
Fax using T.38 Yes
CGPN can be suppressed Yes
CLIP no screening Yes (only for Danish numbers)
Reverse Media Negotiation Yes
Long time call possible Yes
External Transfer Yes
NAT Detection Yes
Voice Quality OK? Yes
Redundancy Mechanism Yes
SIP over TCP Yes

Direct Dial In

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

DTMF

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

Hold/Retrieve

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

Transfer with consultation

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

Transfer with consultation (alerting only)

Tested feature Result
Call can be transferred Yes
Held end hears music on hold or dialling tone Yes
Call returns to transferring device if the third Endpoint is not available Yes

Blind Transfer

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

Broadcast Group & Waiting Queue

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

Configuration

Firmware version

All innovaphone devices use V9hf15 as firmware.

SIP - Trunk

Here's the configuration of the SIP gateway interface.

No need to use STUN, so It's optional.

Fixed AOR should not be used to send different CPGN since only "FROM" header it's used for CGPN and not Prefered-ID field on SIP Invite.

Number Mapping

Route Settings

Uni-tel, as most SIP Providers do, doesn't support overlap sending. You must enable the enblock sending of the phone number. You can do this by enabling Force enblock in the appropriate outgoing Route.

The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this check-box is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.

Redundancy

Redundancy could be done by DNS SRV.

When resolving the proxy name the provider may give us several IP address in the SRV record. We select one address and keep the others as fallback. Methods used to select one address of several returned addresses are also used for load balancing. Whether load balancing and/or redundancy is implemented only depends on priority values of the returned addresses.

Known Issues

Retrieved from "https://wiki.innovaphone.com/index.php?title=Howto:Uni-tel_SIP_Provider_Compatibility_Test&oldid=26111"