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

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{{FIXME|reason= This is a temporary version of innovaphone SIP Interop Tests. Merge back to official wiki when finished}}
== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=January 14th, 2022|url=https://www.openip.fr/en/telephony-en/produits-telephonie-sip-trunk-touch/|productname=SIP_Trunk_Touch|providername=OpenIP}}
=== Remarks ===
*This provider use a custom value of the Identity header. You must change the automated <nowiki> "add UUI" </nowiki> configuration in route map.
*The UUI should be a SIP URI with a valid number present on your trunk.
*See detail on wiki: [http://wiki.innovaphone.com/index.php?title=Howto:How_to_customize_the_From/Identity_header_value_at_SIP_interfaces How_to_customize_the_From/Identity_header_value_at_SIP_interfaces]


{{FIXME|reason= Article name must have format Country_-_Provider_-_Productname_-_SIP Provider (YYYY)}}
<internal>Provider SBC: </internal>
== 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.


Issues found were:
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
; Clip No Screening
{{SIP_TEST_ISSUES_MR_INTRO}}
; Fax T38
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; SRTP
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; Redundancy Failover
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; Provider only offers 1 codec (G711U or G711A)
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}


<small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[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_INCOMING_FAILS|CONN_NR_INCOMING]], [[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_ONNET_FAILS|FAX_T38_ONNET]], [[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_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_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[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_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[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>


For more details about this issues, see the respective test-results sections.
== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}


== Current test state ==
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}


<!--{{Template:Compat Status "planned"}} -->
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
{{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 23th, 2015. -->
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
<!-- [[Category:RecProd|{{PAGENAME}}]] -->
<!-- [[Category:3rdParty SIP Provider|{{PAGENAME}}]] -->


== Tests with MediaRelay ==
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}
; 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
; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}


; CLIR : OK
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}


; Clip No Screening(CLNS) : CLNS is not possible
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}
: 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>)


; 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).
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}


; 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.
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_yes_not_recommended}}


; SRTP : [<code>text</code>]
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
* WORKING_SRTP
** if WORKING_SRTP is none, use text: The provider does not support audio encryption using SRTP.
** otherwise, use text: The provider supports audio encryption using SRTP for ''value'' of WORKING_SRTP (e.g. ... for incoming, outgoing, intern) calls.


; DTMF (RFC2833) : [<code>text</code>]
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
* WORKING_DTMF_RFC2833
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}
** if yes, use text: OK
** if no, use text: Transmitting DTMF-tones as RTP-events was not possible. This will cause problems with 3rd party devices, since most endpoints send & expect DTMF tones as defined in RFC2833.


; NAT Traversal : [<code>text</code>]
; Codecs : supported to/from PSTN: G711A, G711U and G729
*check WORKING_NAT_SCHEMES
: supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
** if WORKING_NAT_SCHEMES contains C & A, use text: The provider detects clients behind NAT and can handle calls to them without requiring the clients to use NAT-traversal methods like STUN.
*** WORKING_NAT_SCHEMES contains only A but no C, use text above. Add comment: However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call.
** if WORKING_NAT_SCHEMES contains D & B, but (no C or A), use text: The provider cannot handle calls to clients behind NAT. Clients are requires to use NAT-traversal methods like STUN. Drawback of this solution is that STUN doesn't work for all NAT routers (i.e. routers doing symmetric NAT). Because of this limitation, it depends on the customer network equipment whether the SIP-tunk is usable or not.
*** WORKING_NAT_SCHEMES contains only B but no (C,A,D) use text above. Add comment: However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call.


; Reverse Media Negotiation : [<code>text</code>]
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
*check WORKING_REV_MEDIA_NEG
** if yes, use text: OK
** if no, use text: Reverse-media negotiation is not supported, MediaRelay and Exclusive coder must be activated on the SIP-interface.


; Mobility Call : [<code>text</code>]
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}
* WORKING_DTMF_SIP_INFO
** if yes, use text: OK
** if no, use text: 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.


; Redundancy : [<code>text</code>]
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}
* WORKING_REDUNDANCY
** if no, use text: Registration of two SIP-interfaces on the same SIP-account is not supported by the provider. As a result, you cannot have a ''Standby'' gateway/PBX using the same account for failover or load-balancing purposes.
** if WORKING_REDUNDANCY=yes, but WORKING_REDUNDANCY_FAILOVER = no, use text: 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.
** if WORKING_REDUNDANCY=yes and WORKING_REDUNDANCY_FAILOVER = yes, use text: Registration of two SIP-interfaces on the same SIP-account is supported by the provider. The provider has a 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 default SIP-registration interval) incoming and outgoing calls might be rejected/fail.
** check USE_TTL in providerdata.h. If not default(i.e 120 sec.), replace 2 minutes in above test with its value.


; IP-Fragmentation : [<code>text</code>]
; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}
*check WORKING_FRAGMENTATION:
** if yes, use text: OK
** if no, use text: IP-Fragmentation is not supported by the provider. When using UDP as Transport protocol, this might cause problem since the fragmentation of the packets cannot be influenced by the sender (PBX), but depends on the routers (IP-hops) to the SIP-provider. The result will be failed calls.


; Large SIP messages : [<code>text</code>]
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
*check WORKING_LARGE_MESSAGES:
** if yes, use text: OK
** if no, use text: Large SIP messages (> 1500 bytes) are not supported by the provider. This might lead to sporadic failure of outbound calls, e.g. if the call has redirection information and by additional data the singling message gets to large for the SIP-provider.


; Correct signalling of Ringing-state : [<code>text</code>]
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
*check WORKING_RINGING:
** if yes, use text: OK.
** if no, use text: 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.


; Early-Media : [<code>text</code>]
* check WORKING_EARLY_MEDIA
** if yes, use text: The provider supports early-media for outbound calls to the PSTN.
** if no, use text: The provider does not support early-media for outbound calls to the PSTN. As a result, a caller will not hear announcement of an PSTN-provider (e.g. The number you dialled does not exist.) or custom ring-tones (e.g. if making an international call)


; Session Timer : [<code>text</code>]
* WORKING_EXPIRES
* is yes, use text: 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.
* if no, use text. The tests regarding the SIP-session timer were not successful. This will result in unwanted call termination on calls exceeding a certain time (default 30 minutes). Because of this, further tests were aborted. [[Benutzer:Sga|Sebastian Gabris]] 11:15, 14. Jan. 2016 (CET) wäre ein KO kriterium. Unklar jedoch ob man das überhaupt erwähnen soll-
; Call Transfer : [<code>text</code>]
* check WORKING_CALL_TRANSFER:
** if it contains <code>consconn consalert blind extern</code>, use text: OK
*** otherwise:
**** if <code>consconn</code> is missing, use text: Transfer with consultation was not tested successfully.
**** if <code>consalert</code> is missing, use text: A semi-attended transfer (i.e. without waiting for the consultation call to be answered) was not tested successfully.
** if it contains <code>blind</code>, use text: A blind transfer (i.e. without a consultation call) was not tested successfully.
**** if <code>extern</code> is missing, use text: An external transfer, with consultation call, was not tested successfully. As a result, it is not possible to transfer an external call (i.e. involving a PSTN participant) back to the PSTN.
[[Benutzer:Sga|Sebastian Gabris]] 12:01, 14. Jan. 2016 (CET)sollte es ein comment in providerdata.h geben, um zu sagen was genau nicht funktioniert hat?
== Tests without MediaRelay==
Listed here are only the test-results that differ from the tests with MediaRelay. If no test were done because MediaRelay is required, use text: The tests without MediaRelay were aborted, since it is required by the provider. [reason] e.g: The reason for it, are audio problems when changing the remote RTP-endpoint during a call and missing support for reverse-media negotiation.
<!-- only needed for tests without nightly-test execution
==Firmware version==
All innovaphone devices use V11r2 SRx as firmware.
-->


==Configuration==
==Configuration==
* Use profile ''Profile-Name'' in the Gateway/Interfaces/SIP menu.
Use profile ''FR-OpenIP-SIP_Trunk_Touch'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
 
: {{SIP_TEST_V13_HINT}}
===Additional Configuration===
* if no point below applies, don't use the Header <nowiki>===Additional Configuration===</nowiki>
* check WORKING_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC
:: if yes, check if G.711A is in WORKING_CODECS use text: FAX requires exclusive codec. Enable G711A Exclusive on all Fax-endpoints.
:: if yes but only G.711U is in WORKING_CODECS use text: FAX requires exclusive codec. Enable G711U Exclusive on all Fax-endpoints.
 
* if WORKING_NAT_SCHEMES contains D but no C, then the provider can work without MR but requires STUN on all endpoints
:: use text: If MediaRelay is not enabled on the SIP-Trunk, all RTP-endpoints must have a STUN-server configured.
 
==Contact==


<!-- [http-link contact form of provider]-->
== Disclaimer ==
{{SIP_TEST_PREFACE}}


<!--[[Category:Compat|{{PAGENAME}}]] -->
[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Latest revision as of 12:50, 20 January 2022

Summary

Tests for the SIP_Trunk_Touch SIP trunk product of the provider OpenIP were completed. Test results have been last updated on January 14th, 2022. Check the history of this article for the date of the first publication of the testreport.

Remarks

<internal>Provider SBC: </internal>

List of Issues found in media-relay Configuration

FAX T38
The provider does not fully support T.38 fax
FAX T38ANDAUDIO
The provider does not support fallback to audio-fax if T.38 fails.
REDIR 302
The provider does not support external call redirection using the SIP 302 Redirect response
SIP INFO
The provider does not support conveying DTMF using the SIP-INFO method.

Here is the list of test-cases that have been performed for this provider: SUBSCRIBER_NR, 180_RINGING, BASIC_CALL, CLIR, CLNS_ONNET, CLNS, CONN_NR_DIFF, CONN_NR_INCOMING, CONN_NR, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38_ONNET, G711A_ONNET, G711A, G711U_ONNET, G711U, G722_ONNET, G729_ONNET, G729, HOLD_RETRIEVE, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, FAX_T38, FAX_T38ANDAUDIO, G722, MOBILITY, OPUS_NB, OPUS_WB, RALERT_DISC, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, REVERSE_MEDIA, SDP_ICE, SDP_RTCP_MUX, SDP_VIDEO, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS

Test Results

This SIP provider requires a media-relay configuration. That is, all media traffic between the SIP provider and all endpoints must flow through the SBC. For this reason, a configuration without media-relay has not been tested and hence no test results for this configuration are listed.

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. However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call.
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.
Dialing of Subscriber Numbers
The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
Correct signalling of Ringing-state
OK
CLIR
OK
Clip No Screening (CLNS)
Straight clip no screening (i.e. sending a foreign number as calling line id) works fine. As this works anyway, it does not matter, if the provider supports the interpretation of History-Info: or Diversion: SIP headers for providing the correct calling line id for diverted calls.
COLP
Outbound and inbound calls to/from the PSTN show the correct connected number.
For outbound calls to the PSTN, an update of the connected number is signalled to the caller.
However, during our test other interop problems were discovered when the Interworking Flag is enabled. As a result, the update of the connected number cannot be signalled.
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.
Transport of faxes using T.38 worked to onnet destinations. However it failed to PSTN destination and moreover fallback to audiofax failed also.
As a result, T.38 is disabled on the SIP-interface and the use of audio-fax is necessary.
Codecs
supported to/from PSTN: G711A, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
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.
SRTP
The provider does not support audio encryption using SRTP.
Call Transfer
OK


Configuration

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

A most recent v13r3 firmware is required to use this SIP-profile. For hints regarding upgrade to v13r3, see Howto:V13_Firmware_Upgrade_V13r2_V13r3

New profiles are added in the course of our V13R3 software Service Releases, see Reference13r3:Release Notes Firmware. Here is an up to date list of tested SIP providers.

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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