Howto:FR - OpenIP - SIP Trunk Touch SIP-Provider (2016): Difference between revisions

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== Summary ==
== Summary ==  
The tests for the [http://www.openip.fr/telephonie-sip-trunk/offre SIP Trunk] of the provider [http://www.openip.fr/ OpenIP] were completed.
{{Template:SIP_TEST_STATUS_ongoing|update=September 13th, 2016|url=|productname=SIP_Trunk|providername=OpenIP}}
<internal>Provider SBC: OpenVoice-15</internal>


Issues found were:
; Clip No Screening
; Fax T38
; SRTP
; Redundancy Failover
; Provider only offers 1 codec (G711U or G711A)


=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; G711A ONNET : {{SIP_TEST_FACT_G711A ONNET}}
; G711A : {{SIP_TEST_FACT_G711A}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; RALERT DISC : {{SIP_TEST_FACT_RALERT DISC}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
; XFER CONS ALERT : {{SIP_TEST_FACT_XFER CONS ALERT}}
: {{SIP_TEST_FACT__unreliable}}


For more details about this issues, see the respective test-results sections.
<small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]</small>


== Current test state ==
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
; FAX AUDIO : {{SIP_TEST_FACT_FAX AUDIO}}
: {{SIP_TEST_FACT__unreliable}}
; XFER CONS ALERT : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}}


<!--{{Template:Compat Status "planned"}} -->
<!-- {{Template:Compat Status "in progress"}} -->
<!-- {{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_"referral_prod."-no-certificate}} -->


Testing of this product has been finalized June 10th, 2016.
== Test Results ==
[[Category:RecProd|{{PAGENAME}}]]
{{SIP_TEST_TESTRESULT_BOTH_INTRO}}
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]
=== {{SIP_TEST_RESULTS_NO_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}
 
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}
 
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
 
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
 
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}
 
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}
:{{Template:SIP_Profile_Test_RALERT_DISC_no}}
 
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}
 
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}
 
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
 
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no}}
 
; Codecs : supported to/from PSTN: G711U
: supported onnet (VoIP to VoIP): G711U
 
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
 
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}
 
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}
 
; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_without_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}} {{Template:SIP_Profile_Test_MobilityCall_no_clns_no_history_or_diversion}}


== Tests with MediaRelay ==
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
; Registration : The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted. Moreover it requires all involved network elements to support IP-fragmentation.


; CLIP : OK
; Call Transfer :  
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consalert}}


; CLIR : OK


; Clip No Screening(CLNS) : CLNS is not possible
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
: However, redirection based on the Diversion number or based on "302 Moved Temporary" is not possible. As a result, forwarded calls or Mobility calls will not see the original calling-number but the diverting parting number as caller(if the TrunkLine - object is configured with <code>Set Calling=Diverting No</code>)
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}


; Codecs : The provider support the following codecs: G711u, G711a (However only one can be used at time, it's necessary to ask to switch codec).
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}


; Fax : Transport of faxes to/from the PSTN via G.711(A/U) codec was tested successfully. However transport of faxes to/from the PSTN using the T.38 protocol was not tested successfully. This is important for the innovaphone Fax-server. Since the provider doesn't support T.38, you must use the "Audiofax" feature(i.e 2 DSPs) for each fax-call done by the innovaphone Fax-interface.
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}


; SRTP : The provider does not support audio encryption using SRTP.
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}


; DTMF (RFC2833) : OK
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}


; NAT Traversal : The provider detects clients behind NAT and can handle calls to them without requiring the clients to use NAT-traversal methods like STUN.
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}
:{{Template:SIP_Profile_Test_RALERT_DISC_no}}


; Reverse Media Negotiation : OK
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}


; Mobility Call : Transmitting DTMF-tones as SIP-INFO messages was not possible, however we can do Mobility-calls since we use MediaRelay on the SIP-Interface.
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}


; Redundancy : Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for a duration of up to 2  minutes (i.e registration interval) incoming and outgoing calls might be rejected/fail.
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}


; IP-Fragmentation : OK
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_no}}


; Large SIP messages : OK
; Codecs : supported to/from PSTN: G711U
: supported onnet (VoIP to VoIP): G711U


; Correct signalling of Ringing-state : Ringing not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing. <br>Additionally external callers forwarded/transferred back to the PSTN, will get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call.
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}


; Early-Media : The provider supports early-media for outbound calls to the PSTN.
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}


; Session Timer : The tests regarding the SIP-session timer were successful. This means that either no session expiry is used or that it is used and works. It does not imply that session expiry actually is used.
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}


; Call Transfer : OK
; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}} {{Template:SIP_Profile_Test_MobilityCall_no_clns_no_history_or_diversion}}


== Tests without MediaRelay==
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
Listed here are only the test-results that differ from the tests with MediaRelay.


; Mobility Call : Transmitting DTMF-tones as SIP-INFO messages was not possible, which is required in-order to use Mobility-calls without MediaRelay on the SIP-Interface.
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}


<!-- only needed for tests without nightly-test execution
==Firmware version==
All innovaphone devices use V11r2 SRx as firmware.
-->


==Configuration==
==Configuration==
* Use profile ''FR-OpenIP-SIP_Trunk'' in the Gateway/Interfaces/SIP menu.
Use profile ''FR-OpenIP-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.


==Contact==
== Disclaimer ==
{{SIP_TEST_PREFACE}}


[http://www.openip.fr/contact contact page of provider]
[[Category:Compat|{{PAGENAME}}]]
 
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]
<!--[[Category:Compat|{{PAGENAME}}]] -->

Revision as of 14:01, 13 September 2016

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated September 13th, 2016) and may (and probably will) change. <internal>Provider SBC: OpenVoice-15</internal>


List of Issues found in no media-relay Configuration

This is a list of all issues found in a configuration where the media stream between endpoints and the SIP provider - as opposed to the signalling - is not routed through the SBC.

180 RINGING
The provider does not send a 180 Ringing response when the called party alerts.
CLNS ONNET
Onnet-Calls (that is, within the provider's network) do not allow foreign calling party numbers (CGPN). In other words, clip no screening is not possible for on-net calls.
CLNS
Outgoing calls cannot be sent with a foreign calling party number (CLI).
FAX T38
The provider does not fully support T.38 fax
G711A ONNET
The provider does not support the G711A codec for on-net calls.
G711A
The provider does not fully support the G711A codec
MOBILITY
The provider can not send DTMF signals via SIP-INFO messages.
RALERT DISC
Call disconnected by far end during alert does not disconnect locally
REDIR 302
The provider does not support external call redirection using the SIP 302 Redirect response
REDIR DIVHDR
The provider does not support maintaining the original caller ID for redirected calls (e.g. CFU) when the original caller is present in a Diversion: header.
REDIR HISTHDR
The provider does not support maintaining the original caller ID for redirected calls (e.g. CFU) when the original caller is present in a History-Info: header.
SIP INFO
The provider does not support conveying DTMF using the SIP-INFO method.
XFER CONS ALERT
The provider does not fully support consultation call transfer after alert scenarios.
Our tests of this feature have shown unstable results (that is, the feature sometimes worked and sometimes did not). This may for example be caused by different equipment used at the provider side (e.g. media gateways) which behave differently.

Here is the list of test-cases that have been performed for this provider: 180_RINGING, BASIC_CALL, CLIR, CLNS_ONNET, CLNS, CONN_NR_DIFF, CONN_NR, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38, G711A_ONNET, G711A, G711U_ONNET, G711U, G722_ONNET, G722, G729_ONNET, G729, HOLD_RETRIEVE, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, MOBILITY, OPUS_NB, OPUS_WB, RALERT_DISC, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, REVERSE_MEDIA, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS

List of Issues found in media-relay Configuration

This section lists the results that differ from the results for the first configuration.

FAX AUDIO
The provider does not fully support Audiofax (i.e. non-T.38)
Our tests of this feature have shown unstable results (that is, the feature sometimes worked and sometimes did not). This may for example be caused by different equipment used at the provider side (e.g. media gateways) which behave differently.
XFER CONS ALERT
This feature, which is unstable in the first configuration, works fine in the second configuration.


Test Results

This section explains the test results for all possible configurations in more detail.

Configuration without media-relay

Registration
The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted(TLS). Moreover it requires all involved network elements to support IP-fragmentation.
NAT Traversal
The provider detects clients behind NAT and can handle calls to them without requiring the clients to use NAT-traversal methods like STUN.
DTMF (RFC2833)
The provider can convey DTMF digits using the RTP payload method as per RFC2833.
Session Timer
The tests regarding the SIP-session timer were successful. This means that either no session expiry is used or that it is used and works. It does not imply that session expiry actually is used.
Redundancy
Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for 2 minutes incoming and outgoing calls might be rejected/fail.
Correct signalling of Ringing-state
Ringing is not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing.
Additionally external callers forwarded/transferred back to the PSTN, may get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call. As a result, Carrier w/o Alerting is required in all PBX 'Mobility' objects.
An outgoing call that is disconnected by the far end during alert is not disconnected locally. Typically, the provider may play a message to the effect that the call has been rejected or the call can not be completed so that the calling user will hang up. This may be OK, however, it may create issues with automated calls (e.g. fax or modem) which do not listen to the announcement and keep waiting for the far end to accept the call.
CLIR
OK
Clip No Screening (CLNS)
CLIP no screening (CLNS) is not possible, that is, outgoing calls can only have the own subscriber number (or extension), no foreign numbers as calling line id. This affects for example externally forwarded and mobility calls.
Also, there is no other method available for this provider to make sure externally forwarded calls will show proper calling line identification (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the forwarded call will at least carry the diverting party number (DGPN) as calling party number (CGPN/CLI) when alerting at the forwarded-to target. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
Early-Media
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
Fax
Transport of faxes to/from the PSTN via G.711 codec was tested successfully.
The provider does not support T.38 fax calls to the PSTN. You need to use audio-fax therefore.
Codecs
supported to/from PSTN: G711U
supported onnet (VoIP to VoIP): G711U
IP-Fragmentation
OK
Large SIP messages
OK
Reverse Media Negotiation
OK
Mobility Calls
Transmitting DTMF-tones as SIP-INFO messages is not supported. In a no-media-relay configuration, DTMF signalling can thus not be conveyed to the PBX. Mobility calls will not work.
As the provider does not provide proper alert signalling, you will need to configure the Carrier w/o Alerting check-mark in the PBX Mobility object.
Neither Clip no screening (CLNS) nor call redirection using the SIP Diversion: or History-Info is supported. Calls forwarded to mobility devices will thus not have the original callers calling line id (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the mobility call will at least carry the mobility user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
SRTP
The provider does not support audio encryption using SRTP.
Call Transfer
The provider does not handle internally transferred-after-alert calls.


Configuration with media-relay

Registration
The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted(TLS). Moreover it requires all involved network elements to support IP-fragmentation.
NAT Traversal
The provider detects clients behind NAT and can handle calls to them without requiring the clients to use NAT-traversal methods like STUN.
DTMF (RFC2833)
The provider can convey DTMF digits using the RTP payload method as per RFC2833.
Session Timer
The tests regarding the SIP-session timer were successful. This means that either no session expiry is used or that it is used and works. It does not imply that session expiry actually is used.
Redundancy
Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for 2 minutes incoming and outgoing calls might be rejected/fail.
Correct signalling of Ringing-state
Ringing is not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing.
Additionally external callers forwarded/transferred back to the PSTN, may get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call. As a result, Carrier w/o Alerting is required in all PBX 'Mobility' objects.
An outgoing call that is disconnected by the far end during alert is not disconnected locally. Typically, the provider may play a message to the effect that the call has been rejected or the call can not be completed so that the calling user will hang up. This may be OK, however, it may create issues with automated calls (e.g. fax or modem) which do not listen to the announcement and keep waiting for the far end to accept the call.
CLIR
OK
Clip No Screening (CLNS)
CLIP no screening (CLNS) is not possible, that is, outgoing calls can only have the own subscriber number (or extension), no foreign numbers as calling line id. This affects for example externally forwarded and mobility calls.
Also, there is no other method available for this provider to make sure externally forwarded calls will show proper calling line identification (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the forwarded call will at least carry the diverting party number (DGPN) as calling party number (CGPN/CLI) when alerting at the forwarded-to target. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
Early-Media
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
Fax
Audio-Fax calls (that is, fax calls without T.38) do not work.
The provider does not support T.38 fax calls to the PSTN. You need to use audio-fax therefore.
Codecs
supported to/from PSTN: G711U
supported onnet (VoIP to VoIP): G711U
IP-Fragmentation
OK
Large SIP messages
OK
Reverse Media Negotiation
OK
Mobility Calls
Transmitting DTMF-tones as SIP-INFO messages is not supported, however mobility calls are still possible as in the media-relay configuration, the SBC will convey DTMF signalling to the PBX.
As the provider does not provide proper alert signalling, you will need to configure the Carrier w/o Alerting check-mark in the PBX Mobility object.
Neither Clip no screening (CLNS) nor call redirection using the SIP Diversion: or History-Info is supported. Calls forwarded to mobility devices will thus not have the original callers calling line id (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the mobility call will at least carry the mobility user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
SRTP
The provider does not support audio encryption using SRTP.
Call Transfer
OK


Configuration

Use profile FR-OpenIP-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

Disclaimer

These tests look at a number of interoperability scenarios between innovaphone SIP devices and a given SIP trunk product. As we are enhancing our testing procedures, nature and number of these tests will vary.

All test results document the fact how the tested combination performs in the tested scenario. It explicitly does not comment on the question what the reason is for the behaviour nor if and how it could be changed. It thus does not imply that either the SIP trunk provider or the innovaphone device fails in any way. It merely says that the combination does not perform as defined by the test.

If not mentioned otherwise, all tests are repeated on a weekly basis (a.k.a. nightly tests) using the then-current innovaphone firmware. If a tested combination keeps performing differently from the state documented here during nightly tests, we reserve the right to update this article accordingly.

Some tests do not have consistent results. This may occur for various reasons. If so, we document the tests result as being unstable.

Some of the tested SIP trunk products are not available for nightly tests. In this case, the fact will be noted in the Summary section.

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