Howto:FR - OVH Telecom - SIP Trunk SIP-Provider (2016): Difference between revisions

From innovaphone wiki
Jump to navigation Jump to search
No edit summary
Line 1: Line 1:
== Summary ==
== Summary ==  
The tests for the [https://www.ovhtelecom.fr/telephonie/sip-trunk/ SIP-trunk] of the provider [https://www.ovhtelecom.fr/ OVH Telecom] were completed successfully.
The tests for the ''[https://www.ovhtelecom.fr/telephonie/sip-trunk SIP_Trunk]'' SIP trunk product of the provider ''OVH_Telecom'' were completed.


Issues found were:
Testing of this product has been finalized August 11st, 2016.
; Clip No Screening
; Fax
; SRTP
; Mobility Call
; Redundancy
; Correct signalling of Ringing-state


=== Remarks ===
08/2016: We are still investigating the Audio Fax failures (see ''FAX AUDIO'' below). <internal>Mantis #127677</internal>


For more details about this issues, see the respective test-results sections.
=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; DTMF : {{SIP_TEST_FACT_DTMF}}
: {{SIP_TEST_FACT__unreliable}}
; FAX AUDIO : {{SIP_TEST_FACT_FAX AUDIO}}
: {{SIP_TEST_FACT__unreliable}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; RALERT DISC : {{SIP_TEST_FACT_RALERT DISC}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}


== Current test state ==
<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_FAILS|CLNS]], [[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>


<!--{{Template:Compat Status "planned"}} -->
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
<!-- {{Template:Compat Status "in progress"}} -->
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}}
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
{{SIP_TEST_ISSUES_MR_INTRO}}
<!-- {{Template:Compat_Status_"engineered_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
; MOBILITY : {{SIP_TEST_FACT_WORKSINALTERNATE_NOT_IN_PRIMARY}}
<!-- {{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 May 24th, 2016.
[[Category:RecProd|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]


== Tests with MediaRelay ==
== Test Results ==
; 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.
{{SIP_TEST_TESTRESULT_BOTH_INTRO}}
=== {{SIP_TEST_RESULTS_NO_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}
 
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}} {{Template:SIP_Profile_Test_DTMF_unreliable}}
 
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
 
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}
 
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}
 
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}
:{{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_no_clns_302_recommended}}
 
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
 
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no}}
 
; Codecs : supported to/from PSTN: G711A, G711U and G729
: supported onnet (VoIP to VoIP): G711A, G711U and G729
 
; 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 Call :  {{Template:SIP_Profile_Test_MobilityCall_no_without_MediaRelay}}
 
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
 
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}


; CLIP : OK


; CLIR : OK
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}


; Clip No Screening(CLNS) : CLNS is not possible, however Redirection using "302 Moved Temporary" is possible. In case of a call forward, the incoming call is redirected to the PSTN. From point of view of the PBX the call doesn't exist after the redirection and is handled by the SIP-Provider. This is equivalent to "Partial Rerouting/Call Deflection" in ISDN. When a redirection is done, the redirection target will receive the original calling number, so for forwarded calls this will have the same effect as CLNS. However this redirection is not usable for Mobility calls or plain CLNS-calls. A plain CLNS-call is an outbound call with a "wrong" CGPN number (i.e. one that does not belong to the trunk). This happens e.g. if a call-center wants to send a service number (such 0800xxx) as CLI.
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}} {{Template:SIP_Profile_Test_DTMF_unreliable}}
:To use call redirection with ''302 Moved temporary'', you have to enable ''Reroute supported'' at your [[Reference10:PBX/Objects/Trunk_Line | TrunkLine]] PBX-object. Additionally you must enable ''Interworking'' on the inbound [[Reference9:Gateway/Routes/Map | route]] from the Provider to the PBX(i.e. SIPx -> RSx).


; Codecs : The provider support the following codecs: G711A, G711U, G729A. The following codecs are not supported: G722, Opus.
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_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.
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}


; SRTP : The provider does not support audio encryption using SRTP.
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}


; DTMF (RFC2833) : OK
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}
:{{Template:SIP_Profile_Test_RALERT_DISC_no}}


; 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.
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}


; Reverse Media Negotiation : OK
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_no_clns_302_recommended}}


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


; 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.
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no}}


; IP-Fragmentation : OK
; Codecs : supported to/from PSTN: G711A, G711U and G729
: supported onnet (VoIP to VoIP): G711A, G711U and G729


; Large SIP messages : OK
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}


; 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.
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}


; Early-Media : The provider supports early-media for outbound calls to the PSTN.
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_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.
; Mobility Call : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}


; Call Transfer : OK
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}


== Tests without MediaRelay==
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
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.


==Configuration==
==Configuration==
* Use profile (e.g. ''FR-OVH_Telecom-SIP_Trunk'') in the Gateway/Interfaces/SIP menu.
Use profile ''FR-OVH_Telecom-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
 
==Contact==


[https://www.ovh.com/fr/support/ contact form of provider]
== Disclaimer ==
{{SIP_TEST_PREFACE}}


[[Category:Compat|{{PAGENAME}}]]
[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Revision as of 16:23, 11 August 2016

Summary

The tests for the SIP_Trunk SIP trunk product of the provider OVH_Telecom were completed.

Testing of this product has been finalized August 11st, 2016.

Remarks

08/2016: We are still investigating the Audio Fax failures (see FAX AUDIO below). <internal>Mantis #127677</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.

CLNS
Outgoing calls cannot be sent with a foreign calling party number (CLI).
DTMF
The provider does not fully support reliable transportation of DTMF signals (DTMF tones are treated separately from voice data). There may be different symptoms like no DTMF at all, no DTMF at the beginning of a call, loss of some DTMF digits in a multi-digit DTMF sequence, duplication of DTMF digits or DTMF digits echoed back to the sender
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.
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.
FAX T38
The provider does not fully support T.38 fax
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 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.

Here is the list of test-cases that have been performed for this provider: 180_RINGING, BASIC_CALL, CLIR, CLNS, 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.

MOBILITY
This feature, which does not work 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.
DTMF (RFC2833)
The provider can convey DTMF digits using the RTP payload method as per RFC2833. However, DTMF handling overall does not work reliably.
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.
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.
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
OK
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. However, call redirection using the SIP 302 Redirect header is supported by the provider. Rerouting can be enabled by setting the Reroute supported check-mark in the corresponding Trunk object configuration. Also, the Interworking check-mark must be set in the route used for calls from and to the SIP provider. We recommend to use this option as this will send proper calling line identification (CLI) for externally redirected calls. Mobility calls(i.e. calls to a mobility destination) or CFBs/CFNRs are not affected by this redirection and will not show the CLI of the original caller or be rejected by the provider. Therefore, we recommend to additionally set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the calls will at least carry the user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device/forwarding destination.
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. However, all fax endpoints must be configured with exclusive codec "G711A".
The provider does not support T.38 fax calls to the PSTN. You need to use audio-fax therefore.
Codecs
supported to/from PSTN: G711A, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U and G729
IP-Fragmentation
OK
Large SIP messages
OK
Reverse Media Negotiation
OK
Mobility Call
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.
SRTP
The provider does not support audio encryption using SRTP.
Call Transfer
OK


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.
DTMF (RFC2833)
The provider can convey DTMF digits using the RTP payload method as per RFC2833. However, DTMF handling overall does not work reliably.
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.
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.
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
OK
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. However, call redirection using the SIP 302 Redirect header is supported by the provider. Rerouting can be enabled by setting the Reroute supported check-mark in the corresponding Trunk object configuration. Also, the Interworking check-mark must be set in the route used for calls from and to the SIP provider. We recommend to use this option as this will send proper calling line identification (CLI) for externally redirected calls. Mobility calls(i.e. calls to a mobility destination) or CFBs/CFNRs are not affected by this redirection and will not show the CLI of the original caller or be rejected by the provider. Therefore, we recommend to additionally set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the calls will at least carry the user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device/forwarding destination.
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. However, all fax endpoints must be configured with exclusive codec "G711A".
The provider does not support T.38 fax calls to the PSTN. You need to use audio-fax therefore.
Codecs
supported to/from PSTN: G711A, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U and G729
IP-Fragmentation
OK
Large SIP messages
OK
Reverse Media Negotiation
OK
Mobility Call
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-OVH_Telecom-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.

Retrieved from "https://wiki.innovaphone.com/index.php?title=Howto:FR_-_OVH_Telecom_-_SIP_Trunk_SIP-Provider_(2016)&oldid=43673"