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

From innovaphone wiki
Jump to navigation Jump to search
(New page: == Summary == {{Template:SIP_TEST_STATUS_ongoing|update=August 24th, 2017|url=https://www.vsenet.de/de/geschftskunden/sprache.html|productname=EASY PHONE_NGN SIP_Trunk|providername=VSE NE...)
 
No edit summary
Line 1: Line 1:
== Summary ==  
== Summary ==  
{{Template:SIP_TEST_STATUS_ongoing|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_ongoing|update=August 24th, 2017|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>




=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
; 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}}
Line 18: Line 21:


<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_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>
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
: {{SIP_TEST_FACT__unreliable}}
; BASIC CALL : {{SIP_TEST_FACT_BASIC CALL}}
: {{SIP_TEST_FACT__unreliable}}
; CLIR : {{SIP_TEST_FACT_CLIR}}
: {{SIP_TEST_FACT__unreliable}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
: {{SIP_TEST_FACT__unreliable}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
: {{SIP_TEST_FACT__unreliable}}
; CONN NR INCOMING : {{SIP_TEST_FACT_CONN NR INCOMING}}
: {{SIP_TEST_FACT__unreliable}}
; CONN NR : {{SIP_TEST_FACT_CONN NR}}
: {{SIP_TEST_FACT__unreliable}}
; DTMF : {{SIP_TEST_FACT_DTMF}}
: {{SIP_TEST_FACT__unreliable}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
: {{SIP_TEST_FACT__unreliable}}
; FAX AUDIO : {{SIP_TEST_FACT_FAX AUDIO}}
: {{SIP_TEST_FACT__unreliable}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
: {{SIP_TEST_FACT__unreliable}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
: {{SIP_TEST_FACT__unreliable}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
: {{SIP_TEST_FACT__unreliable}}
; G711A ONNET : {{SIP_TEST_FACT_G711A ONNET}}
: {{SIP_TEST_FACT__unreliable}}
; G711A : {{SIP_TEST_FACT_G711A}}
: {{SIP_TEST_FACT__unreliable}}
; HOLD RETRIEVE : {{SIP_TEST_FACT_HOLD RETRIEVE}}
: {{SIP_TEST_FACT__unreliable}}
; IP FRAGMENTATION : {{SIP_TEST_FACT_IP FRAGMENTATION}}
: {{SIP_TEST_FACT__unreliable}}
; LARGE SIP MESSAGES : {{SIP_TEST_FACT_LARGE SIP MESSAGES}}
: {{SIP_TEST_FACT__unreliable}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
: {{SIP_TEST_FACT__unreliable}}
; RALERT DISC : {{SIP_TEST_FACT_RALERT DISC}}
: {{SIP_TEST_FACT__unreliable}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
: {{SIP_TEST_FACT__unreliable}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
: {{SIP_TEST_FACT__unreliable}}
; SUBSCRIBER NR : {{SIP_TEST_FACT_SUBSCRIBER NR}}
: {{SIP_TEST_FACT__unreliable}}
; XFER BLIND : {{SIP_TEST_FACT_XFER BLIND}}
: {{SIP_TEST_FACT__unreliable}}
; XFER CONS ALERT : {{SIP_TEST_FACT_XFER CONS ALERT}}
: {{SIP_TEST_FACT__unreliable}}
; XFER CONS EXT : {{SIP_TEST_FACT_XFER CONS EXT}}
: {{SIP_TEST_FACT__unreliable}}
; XFER CONS : {{SIP_TEST_FACT_XFER CONS}}
: {{SIP_TEST_FACT__unreliable}}
; SDP ICE : {{SIP_TEST_FACT_SDP ICE}}
: {{SIP_TEST_FACT__unreliable}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
: {{SIP_TEST_FACT__unreliable}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
: {{SIP_TEST_FACT__unreliable}}




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


Line 63: Line 132:
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
; 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}} {{Template:SIP_Profile_Test_DTMF_unreliable}}
; 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_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_no_in_no}}{{Template:SIP_Profile_Test_COLP_diff_no}}
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}
; Codecs : supported to/from PSTN:
: supported onnet (VoIP to VoIP):
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_no}}
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_no}}
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}
; 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}}
; Provider supports dialling subscriber numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}
; Call Transfer :
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consconn}}
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consalert}}
: {{Template:SIP_Profile_Test_CALL_TRANSFER_blind}}
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consext}}




Revision as of 15:25, 24 August 2017

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated August 24th, 2017) and may (and probably will) change.

Remarks

- Tests with Media-Relay were unreliable in DTMF transmissions for that reason all tests were excluded.

<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 T38 ONNET
The provider does not support T.38 fax for onnet calls.
FAX T38
The provider does not fully support T.38 fax
MOBILITY
The provider can not send DTMF signals via SIP-INFO messages.
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.
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.
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.
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.
SIP INFO
The provider does not support conveying DTMF using the SIP-INFO method.
SDP VIDEO
The provider does not support receiving video media capabilities in the SDP-part of a SIP message.

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, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, SUBSCRIBER_NR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, SDP_ICE, SDP_RTCP_MUX, SDP_VIDEO

List of Issues found in media-relay Configuration

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

180 RINGING
The provider does not send a 180 Ringing response when the called party alerts.
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.
BASIC CALL
Template:SIP TEST FACT BASIC CALL
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.
CLIR
The provider does not fully support suppression of the calling line id (CLIR) using the SIP Privacy: Id header.
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.
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.
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.
CLNS
Outgoing calls cannot be sent with a foreign calling party number (CLI).
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.
CONN NR INCOMING
Incoming calls from the PSTN don't show a correct connected number to the calling party.
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.
CONN NR
For outbound calls to the PSTN the provider transmits an incorrect connected number.
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.
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.
EARLY MEDIA INBOUND
The provider does not support early-media (i.e. establish RTP-stream before 200 OK/connect) for calls to the PSTN.
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 ONNET
The provider does not support T.38 fax for onnet calls.
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
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 T38ANDAUDIO
The provider does not support fallback to audio-fax if T.38 fails.
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.
G711A ONNET
The provider does not support the G711A codec for on-net calls.
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.
G711A
The provider does not fully support the G711A codec
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.
HOLD RETRIEVE
The provider does not support Hold/Retrieve of a call.
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.
IP FRAGMENTATION
Template:SIP TEST FACT IP FRAGMENTATION
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.
LARGE SIP MESSAGES
The provider does not support large SIP messages (> 1500 bytes).
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.
MOBILITY
The provider can not send DTMF signals via SIP-INFO messages.
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.
RALERT DISC
Call disconnected by far end during alert does not disconnect locally
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.
REDIR 302
The provider does not support external call redirection using the SIP 302 Redirect response
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.
REVERSE MEDIA
The provider does not support reverse media negotiation (a.k.a. late SDP)
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.
SIP INFO
The provider does not support conveying DTMF using the SIP-INFO method.
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.
SUBSCRIBER NR
The provider does not support dialling numbers in subscriber number format.
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 BLIND
The provider does not fully support blind call transfer 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.
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.
XFER CONS EXT
The provider does not fully support external consultation call transfer 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.
XFER CONS
The provider does not fully support consultation call transfer after connect 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.
SDP ICE
The provider does not support receiving ICE candidates in the SDP-part of a SIP message.
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.
SDP RTCP MUX
The provider does not support receiving a RTCP-MUX attribute in the SDP-part of a SIP message.
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.
SDP VIDEO
The provider does not support receiving video media capabilities in the SDP-part of a SIP message.
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.


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
Transport of faxes to/from the PSTN via G.711 codec was tested successfully.
Transport of faxes using T.38 failed to PSTN and onnet destinations. Fallback to audio-fax worked.
As a result, T.38 is enabled on the SIP-interface, 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.
SRTP
The provider does not support audio encryption using SRTP.
Provider supports dialling 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. 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.
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
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
CLIR didn't work.
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.
COLP
Outbound and inbound calls to/from the PSTN don't show the correct connected number.
A caller from the PBX to the PSTN (or vice-versa) will receive an incorrect connected number that differs from the dialled number. This might lead to the caller cancelling the call.
For outbound calls to the PSTN, an update of the connected number is not signalled to the caller.
Early-Media
The provider does not support early (that is, before connect) media for outbound calls to the PSTN (hence no inbound early media). This may be an issue in cases where such media is played to the caller (e.g. when calling an unavailable mobile phone).
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:
supported onnet (VoIP to VoIP):
IP-Fragmentation
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
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.
Reverse Media Negotiation
Reverse-media negotiation is not supported. Therefore, a media-relay with exclusive coder configuration will be activated on the SIP-interface.
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.
Provider supports dialling subscriber numbers
The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
Call Transfer
The provider does not handle internally transferred-after-connect calls.
The provider does not handle internally transferred-after-alert calls.
The provider does not handle internally blind-transferred calls.
The provider does not handle externally transferred calls.


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.

Retrieved from "https://wiki.innovaphone.com/index.php?title=Howto:DE_-_VSE_NET_-_EASY_PHONE_NGN_SIP_Trunk_SIP-Provider_(2017)&oldid=47665"