Howto:NL - oneCentral - SIP Trunk TLS SIP-Provider (2020): Difference between revisions

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WARNING: WIKI Hints required (in Howto:NL_-_oneCentral_-_SIP_Trunk_TCP_SIP-Provider_(2020))!!
== Summary ==
You must note the following issues in the wiki article!
{{Template:SIP_TEST_STATUS_complete|update=May 14th, 2024|url=https://onecentral.nl/on-premise-stabiliteit/|productname=SIP_Trunk_TLS|providername=oneCentral}}
=== Remarks ===
{{ Template:SIP_TEST_NO_NIGHTLY_TESTS | fw-version = 14r1 Service Release 14 (1410509)}}


  do not forget to mention the following specials:
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
  - If you intend to use SIPS (SIP/TLS) registration, you need to add the ' sip.onecentral.net' certificate to the trust list of your SBC
{{SIP_TEST_ISSUES_MR_INTRO}}
  - Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}


Various manual steps are required.  So now...
<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]], [[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]]</small>


- from the 9.00 repository, get the latest version of $/13r1/ip6010 (best using your vault client)


- with Visual Studio (currently 2013) open ip6010.sln
== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{SIP_Profile_Test_Registration_TLS_NO_UDP_NO_TCP}}


; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}


== Firmware V12r2 Vault todos ==
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
- add/checkin NL-oneCentral-SIP_Trunk_TCP.xsl to $/12r2/ip6010/relay/products/


- checkout/edit $/12r2/ip6010/relay/relay.mak
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
  - add the following line to the rule for obj/relay_httpdata.h:  
    '' products/NL-oneCentral-SIP_Trunk_TCP.xsl \'' (note that the line MUST start with a tab and end with a backslash!)


- checkout/edit $/12r2/ip6010/relay/relay_ifs.xsl
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}
  - locate the '<select name="profile">' line
  - add the following line to the list of providers
    ''<option value="NL-oneCentral-SIP_Trunk_TCP">NL-oneCentral-SIP_Trunk_TCP</option>''
  - make sure the line is inserted in the correct alphabetic order!
- proceed with "final steps" below


; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}


== Firmware 13r1 Vault todos ==
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}
- add/checkin NL-oneCentral-SIP_Trunk_TCP.xsl to $/13r1/ip6010/relay/products/
- add/checkin sip_product_NL-oneCentral-SIP_Trunk_TCP.js to $/13r1/ip6010/relay/products/


- checkout/edit $/13r1/ip6010/relay/products/sip_products.js
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_history_or_diversion}}
  - add the following line to the correct country property "NL":  
    ''{ name: "oneCentral-SIP Trunk TCP", js: "NL-oneCentral-SIP_Trunk_TCP" },'' (note that the line MUST ends with a comma!)
  - make sure the line is inserted in the correct alphabetic order!


- checkout/edit $/13r1/ip6010/relay/relay.mak
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}
  - add the following lines to the rule for obj/relay_httpdata.h:  
    '' products/NL-oneCentral-SIP_Trunk_TCP.xsl \'' (note that the line MUST start with a tab and end with a backslash!)
    '' products/sip_product_NL-oneCentral-SIP_Trunk_TCP.js,SERVLET_STATIC,HTTP_CACHE+HTTP_NOPWD \'' (note that the line MUST start with a tab and end with a backslash!)


- checkout/edit $/13r1/ip6010/relay/relay_ifs.xsl
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
  - locate the '<select name="profile">' line
  - add the following line to the list of providers
    ''<option value="NL-oneCentral-SIP_Trunk_TCP">NL-oneCentral-SIP_Trunk_TCP</option>''
  - make sure the line is inserted in the correct alphabetic order!
- proceed with "final steps" below


; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}


=Final steps=
; Codecs : supported to/from PSTN: G711A
- create a project task and add it to suggested fixes
: supported onnet (VoIP to VoIP): G711A
  - Fix: [Add/Update] oneCentral-SIP Trunk TCP
  - Bereich: Fixes
  - Release: _Fixes - Suggested Fixes
  - Status: Beendet


- save all changes you have done so far
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}


- create an IP6010 build (either locally [preferred, call ''make Firmware_Build''] or using the builder [do not forget to check-in before in this case])
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}
- DRAM this build to Ohm (from C:/sources/builder/src/13r1/ip6010/bin) and test the provider profile (V12 AND V13!)
- if the build is good, check-in all files to vault


- from the test repository, get the latest version of $/test/13r1/relay/sip-profiles/Makefile (best using your vault client)
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}


  - checkout/edit Makefile in $/test/13r1/relay/sip-profiles/
; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}
  - locate the 'TESTEDPROVIDER=line' line
  - add the following line to the list of tested providers:
    '' NL-oneCentral-SIP_Trunk_TCP \'' (note that the line MUST start with a tab!)


- make sure all your changes are checked in if satisfied
; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}


- create the service release documentation
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
  see http://wiki-intern.innovaphone.com/index.php?title=Entwicklung#Service_Release_Dokumentation for details on the development process for service release documentation
  - login to the project tool (use myApps for that)
  - select 'Area' 'Fixes'
  - in the left pane ('Releases') select and open 'Current Releases'
  - select the current Firmware Version (currently this is 'Firmware 13r1')
    you should now be in  "Fixes" Releases/Current Releases/Firmware 13r1
  - navigate to the next Service Release (e.g. '13r1 Service Release 2')
  - Check if there already is a fix 'SIP-Provider Profile NL-oneCentral-SIP_Trunk_TCP'
  - if so, open it.  if not, create a new fix by clicking on the '+ Fix' button
    in the form, fill in:  
    - Fix: 'SIP-Provider Profile NL-oneCentral-SIP_Trunk_TCP'
    - Release: select the next 13r1 service release
    - Status: 'Aktuell'
    - Beschreibung:
      if this is a new profile, then add the words 'New SIP Provider Profile'
      if this is an updated profile, then add the words 'Updated SIP Provider Profile' plus an explanation why it has been updated (in English)
    - Add 2 Document Links:
      - http://mantis.innovaphone.com/view.php?id=237667
      - http://wiki.innovaphone.com/index.php?title=Howto%3ANL_-_oneCentral_-_SIP_Trunk_TCP_SIP-Provider_%282020%29
  - Save the new fix
  - navigate to the 'Aktuelle Fixe' section and click on the symbol for 'Fix in Communote posten'
      in the communote-form, add 'Techserv/Sip-Provider' in 'Themen'
  - navigate to 'Aktuelle Fixes' section of your fix and click on the symbol for 'Fix erledigt'


- create/update the wiki article (Howto:NL_-_oneCentral_-_SIP_Trunk_TCP_SIP-Provider_(2020)) in http://wiki.innovaphone.com/index.php?title=Howto:NL_-_oneCentral_-_SIP_Trunk_TCP_SIP-Provider_(2020)&action=edit
; SRTP : {{Template:SIP_Profile_Test_SRTP_yes}}
  use content of '<profilename>.wiki.txt' to start with
  do not forget to mention the wiki specials mentioned above
- send email to the provider from Mantis and summarize all the test findings
  do not forget to attach ProviderProfile.png and all relevant traces!
  Here is an email template:  


------------------------------ Provider Email ------------------------------
We have concluded our tests. Outcome so far is documented in http://wiki.innovaphone.com/index.php?title=Howto:NL_-_oneCentral_-_SIP_Trunk_TCP_SIP-Provider_(2020):


Issues you may want to look into (we refer to the traces found in attached zip):


  - <explain issues here, mention traces if any>
==Configuration==
Use profile ''NL-oneCentral-SIP_Trunk_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.


We have created a special configuration form four your product (screenshot ProviderProfile.png attached).
Please note the following configuration hints:
Would you please have a look at it and especially review and check the terms we are using in the form?
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
We can change all the terms used in this form to reflect the terms you are using in your communication between you and your customers.
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' sip.onecentral.net1' certificate to the trust list of your SBC</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>


Hope to hear from you soon so we can finish the process.
: {{SIP_TEST_V13_HINT}}
------------------------------ Provider Email ------------------------------


*** Do not forget to attach ProviderProfile.png as well es all relevant traces.
== Disclaimer ==
{{SIP_TEST_PREFACE}}
 
[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Latest revision as of 13:11, 14 May 2024

Summary

Tests for the SIP_Trunk_TLS SIP trunk product of the provider oneCentral were completed. Test results have been last updated on May 14th, 2024. Check the history of this article for the date of the first publication of the testreport.

Remarks

The provider doesn't offer the possibility to test the SIP-trunk regularly and automatically (e.g. before firmware releases). As a result, we do not know if it will still work or will work with different firmware than the one we tested with.

Tested Firmware: 14r1 Service Release 14 (1410509)

List of Issues found in media-relay Configuration

180 RINGING
The provider does not send a 180 Ringing response when the called party alerts.
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.
CLNS
Outgoing calls cannot be sent with a foreign calling party number (CLI).
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.
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
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, SDP_ICE, SDP_RTCP_MUX, SDP_VIDEO, SIP_INFO, SUBSCRIBER_NR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING


Test Results

This SIP provider requires a media-relay configuration. That is, all media traffic between the SIP provider and all endpoints must flow through the SBC. For this reason, a configuration without media-relay has not been tested and hence no test results for this configuration are listed.

Configuration with media-relay

Registration
The provider supports only TLS as transport protocol. In general TLS is preferred to TCP or UDP, 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.
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)
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.
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. 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
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.
Dialing of Subscriber Numbers
The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
Call Transfer
OK
SRTP
The provider supports audio encryption using SRTP for incoming, outgoing and on-net calls.


Configuration

Use profile NL-oneCentral-SIP_Trunk_TLS in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

  • Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects
  • If you intend to use SIPS (SIP/TLS) registration, you need to add the ' sip.onecentral.net1' certificate to the trust list of your SBC
  • Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
A most recent v13r3 firmware is required to use this SIP-profile. For hints regarding upgrade to v13r3, see Howto:V13_Firmware_Upgrade_V13r2_V13r3

New profiles are added in the course of our V13R3 software Service Releases, see Reference13r3:Release Notes Firmware. Here is an up to date list of tested SIP providers.

Disclaimer

These tests look at a number of interoperability scenarios between innovaphone SIP devices and a given SIP trunk product. As we are enhancing our testing procedures, nature and number of these tests will vary.

All test results document the fact how the tested combination performs in the tested scenario. It explicitly does not comment on the question what the reason is for the behaviour nor if and how it could be changed. It thus does not imply that either the SIP trunk provider or the innovaphone device fails in any way. It merely says that the combination does not perform as defined by the test.

If not mentioned otherwise, all tests are repeated on a weekly basis (a.k.a. nightly tests) using the then-current innovaphone firmware. If a tested combination keeps performing differently from the state documented here during nightly tests, we reserve the right to update this article accordingly.

Some tests do not have consistent results. This may occur for various reasons. If so, we document the tests result as being unstable.

Some of the tested SIP trunk products are not available for nightly tests. In this case, the fact will be noted in the Summary section.

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