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

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== Summary ==  
== Summary ==  
{{Template:SIP_TEST_STATUS_complete|update=June 1st, 2017|url=http://openip.fr/telephonie-sip-trunk/offre|productname=SIP_Trunk|providername=OpenIP}}
{{Template:SIP_TEST_STATUS_complete|update=January 14th, 2022|url=https://www.myopenip.fr/en/sip-trunk-touch|productname=SIP_Trunk_Touch|providername=OpenIP}}
=== Remarks ===
=== Remarks ===
{{Template:SIP_TEST_NO_NIGHTLY_TESTS|fw-version= 12r1 Service Release 8 (121106)}}
*This provider use custom value of Identity header. May you should change the automated <nowiki> "add UUI" </nowiki> configuraiton 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]


<internal>Provider SBC: OpenVoice-15</internal>
<internal>Provider SBC: </internal>


=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
 
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
{{SIP_TEST_ISSUES_MR_INTRO}}
: {{SIP_TEST_FACT__unreliable}}
; CLIR : {{SIP_TEST_FACT_CLIR}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; 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 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}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
; XFER CONS ALERT : {{SIP_TEST_FACT_XFER CONS ALERT}}
: {{SIP_TEST_FACT__unreliable}}
; SUBSCRIBER NR : {{SIP_TEST_FACT_SUBSCRIBER NR}}


<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_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[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]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]]</small>
<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>
 
=== {{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}}




== Test Results ==
== Test Results ==
{{SIP_TEST_TESTRESULT_BOTH_INTRO}}
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_NO_MR_TITLE}} ===
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}


; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}  
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}  


; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
Line 50: Line 32:
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}


; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}
; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}
:{{Template:SIP_Profile_Test_RALERT_DISC_no}}


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


; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}


; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_no}}{{Template:SIP_Profile_Test_COLP_diff_no}}
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}
 
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_yes_not_recommended}}


; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}


; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}


; Codecs : supported to/from PSTN: G711U
; Codecs : supported to/from PSTN: G711A, G711U and G729
: supported onnet (VoIP to VoIP): G711U
: supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729


; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
Line 73: Line 56:
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_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}}
; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}
 
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
 
; Call Transfer :
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consalert}}
 
; Provider supports dialling subscriber numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}
 
 
=== {{SIP_TEST_RESULTS_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_no}}
 
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}
 
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_no}}{{Template:SIP_Profile_Test_COLP_diff_no}}
 
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
 
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_yes}}
 
; 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_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}} {{Template:SIP_Profile_Test_MobilityCall_no_clns_no_history_or_diversion}}


; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
Line 123: Line 62:
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}


; Provider supports dialling subscriber numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}




==Configuration==
==Configuration==
Use profile ''FR-OpenIP-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
Use profile ''FR-OpenIP-SIP_Trunk_Touch'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V12_HINT}}
: {{SIP_TEST_V13_HINT}}
 
== Disclaimer ==
== Disclaimer ==
{{SIP_TEST_PREFACE}}
{{SIP_TEST_PREFACE}}

Revision as of 17:14, 14 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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