Howto:NL - XENOSITE - SIP TRUNK SIP-Provider (2017): Difference between revisions

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Test Wiki
== Summary ==
{{Template:SIP_TEST_STATUS_ongoing|update=April 4th, 2017|url=|productname=SIP_TRUNK|providername=XENOSITE}}
<internal>Provider SBC: </internal>
 
 
=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
 
<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]]</small>
 
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
{{SIP_TEST_ISSUES_NO_ALTERNATE_ISSUES}}
 
== Test Results ==
{{SIP_TEST_TESTRESULT_BOTH_INTRO}}
=== {{SIP_TEST_RESULTS_NO_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}
 
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}
 
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
 
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
 
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}
 
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}
 
; 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_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_yes}}
 
; Codecs : supported to/from PSTN: G711A and G729
: supported onnet (VoIP to VoIP): G711A, G711U, G722 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_no}}
 
; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}
 
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
 
; 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_TCP_TLS}}
 
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}
 
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
 
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
 
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}
 
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}
 
; 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_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_yes}}
 
; Codecs : supported to/from PSTN: G711A and G729
: supported onnet (VoIP to VoIP): G711A, G711U, G722 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_no}}
 
; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}
 
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
 
; Provider supports dialling subscriber numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}
 
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
 
 
==Configuration==
Use profile ''NL-XENOSITE-SIP_TRUNK'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V12_HINT}}
== Disclaimer ==
{{SIP_TEST_PREFACE}}
 
[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Revision as of 12:02, 5 April 2017

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated April 4th, 2017) and may (and probably will) change. <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.

180 RINGING
The provider does not send a 180 Ringing response when the called party alerts.
FAX T38
The provider does not fully support T.38 fax
REDIR 302
The provider does not support external call redirection using the SIP 302 Redirect response
REVERSE MEDIA
The provider does not support reverse media negotiation (a.k.a. late SDP)
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, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, 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 UDP, TCP and TLS as transport protocol. The tests were completed using TLS, since it offers encryption of the transmitted SIP-packets.
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 not supported by the provider. As a result, you cannot have a Standby gateway/PBX using the same account for failover or load-balancing purposes.
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.
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 worked to onnet destinations. However it failed to PSTN destination. Fallback to audiofax 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 and G729
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
IP-Fragmentation
OK
Large SIP messages
OK
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.
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 UDP, TCP and TLS as transport protocol. The tests were completed using TLS, since it offers encryption of the transmitted SIP-packets.
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 not supported by the provider. As a result, you cannot have a Standby gateway/PBX using the same account for failover or load-balancing purposes.
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.
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 worked to onnet destinations. However it failed to PSTN destination. Fallback to audiofax 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 and G729
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
IP-Fragmentation
OK
Large SIP messages
OK
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.
SRTP
The provider does not support audio encryption using SRTP.
Provider supports dialling subscriber numbers
OK
Call Transfer
OK


Configuration

Use profile NL-XENOSITE-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.

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