Howto:DE - VSE NET - EASY PHONE NGN SIP Trunk SIP-Provider (2017): Difference between revisions

From innovaphone wiki
Jump to navigation Jump to search
No edit summary
Line 1: Line 1:
== Summary ==  
== Summary ==  
{{Template:SIP_TEST_STATUS_complete|update=August 24th, 2017|url=https://www.vsenet.de/de/geschftskunden/sprache.html|productname=EASY PHONE_NGN SIP_Trunk|providername=VSE NET}}
{{Template:SIP_TEST_STATUS_complete|update=June 14th, 2018|url=https://www.vsenet.de/de/geschftskunden/sprache.html|productname=EASY PHONE_NGN SIP_Trunk|providername=VSE NET}}
=== Remarks ===
- Tests with Media-Relay were unreliable in DTMF transmissions for that reason all tests were excluded.
 
<internal>Provider SBC: </internal>
<internal>Provider SBC: </internal>


Line 9: Line 6:
=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
; FAX AUDIO : {{SIP_TEST_FACT_FAX AUDIO}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
: {{SIP_TEST_FACT__unreliable}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}


<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_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_FAX_T38_FAILS|FAX_T38]], [[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_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[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_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]]</small>
<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_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_FAX_T38_FAILS|FAX_T38]], [[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_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_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[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}}
{{SIP_TEST_ISSUES_NO_ALTERNATE_ISSUES}}


== Test Results ==
== Test Results ==
Line 33: Line 32:
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}


; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout= 600 seconds}}
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout= 600 seconds}}


; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}
Line 41: Line 40:
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}
; 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_no}}
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}


; 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_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}


; Codecs : supported to/from PSTN: G711A
; Codecs : supported to/from PSTN: G711A
Line 59: Line 58:
; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_without_MediaRelay}}
; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_without_MediaRelay}}


; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}
 
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}


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


; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
=== {{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= 600 seconds}}
 
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}
 
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}
 
; 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_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_no}}
 
; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A and G722
 
; 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}}


; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}


; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}




Line 71: Line 106:
Use profile ''DE-VSE_NET-EASY_PHONE_NGN_SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
Use profile ''DE-VSE_NET-EASY_PHONE_NGN_SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V12_HINT}}
: {{SIP_TEST_V12_HINT}}
== Disclaimer ==
== Disclaimer ==
{{SIP_TEST_PREFACE}}
{{SIP_TEST_PREFACE}}

Revision as of 17:03, 14 June 2018

Summary

Tests for the EASY PHONE_NGN SIP_Trunk SIP trunk product of the provider VSE NET were completed. Test results have been last updated on June 14th, 2018. Check the history of this article for the date of the first publication of the testreport. <internal>Provider SBC: </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.

FAX AUDIO
The provider does not fully support Audiofax (i.e. non-T.38)
FAX T38 ONNET
The provider does not support T.38 fax for onnet calls.
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
SDP VIDEO
The provider does not support receiving video media capabilities in the SDP-part of a SIP message.
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_ONNET, CLNS, CONN_NR_DIFF, CONN_NR_INCOMING, CONN_NR, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38_ONNET, FAX_T38, FAX_T38ANDAUDIO, 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, SDP_ICE, SDP_RTCP_MUX, SDP_VIDEO, SIP_INFO, SUBSCRIBER_NR, 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.

The test results for this configuration are the same, however.

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 600 seconds incoming and outgoing calls might be rejected/fail.
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 not signalled to the caller.
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.
Transport of faxes using T.38 failed to PSTN and onnet destinations. Moreover fallback to audio-fax 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
supported onnet (VoIP to VoIP): G711A and G722
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.
Dialing of Subscriber Numbers
OK
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.
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 600 seconds incoming and outgoing calls might be rejected/fail.
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 not signalled to the caller.
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.
Transport of faxes using T.38 failed to PSTN and onnet destinations. Moreover fallback to audio-fax 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
supported onnet (VoIP to VoIP): G711A and G722
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.
Dialing of Subscriber Numbers
OK
Call Transfer
OK


Configuration

Use profile DE-VSE_NET-EASY_PHONE_NGN_SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

A most recent v12r2 firmware or higher is required to use this SIP-profile. For hints regarding upgrade to v12r2, see Howto:Firmware Upgrade V12r1 V12r2

New profiles are added in the course of our V12R2 software Service Releases, see Support:DVL-Roadmap Firmware 12r2. 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.