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

From innovaphone wiki
Jump to navigation Jump to search
No edit summary
No edit summary
Line 1: Line 1:
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=March 30th, 2020|url=https://onecentral.nl/on-premise-stabiliteit/|productname=SIP_Trunk_TLS|providername=oneCentral}}
=== Remarks ===
- 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


  do not forget to mention the following specials:
- Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
  - 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
  - Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'


Various manual steps are required.  So now...
<internal>Provider SBC: Hermes SNS</internal>


- 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
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
: {{SIP_TEST_FACT__unreliable}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
: {{SIP_TEST_FACT__unreliable}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
: {{SIP_TEST_FACT__unreliable}}
; RALERT DISC : {{SIP_TEST_FACT_RALERT DISC}}
; 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}}
; XFER CONS EXT : {{SIP_TEST_FACT_XFER CONS EXT}}
: {{SIP_TEST_FACT__unreliable}}
; XFER CONS : {{SIP_TEST_FACT_XFER CONS}}
: {{SIP_TEST_FACT__unreliable}}


<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_VIDEO_FAILS|SDP_VIDEO]], [[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>


== Firmware V12r2 Vault todos ==
- add/checkin NL-oneCentral-SIP_Trunk_TCP.xsl to $/12r2/ip6010/relay/products/


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


- checkout/edit $/12r2/ip6010/relay/relay_ifs.xsl
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}
  - 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


; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}


== Firmware 13r1 Vault todos ==
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_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
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_yes}}|timeout=2 minutes}}
  - 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
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}
  - add the following lines to the rule for obj/relay_httpdata.h:  
:{{Template:SIP_Profile_Test_RALERT_DISC_no}}
    '' 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
; CLIR : {{Template:SIP_Profile_Test_CLIR_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


; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_history_or_diversion}}


=Final steps=
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}
- create a project task and add it to suggested fixes
  - Fix: [Add/Update] oneCentral-SIP Trunk TCP
  - Bereich: Fixes
  - Release: _Fixes - Suggested Fixes
  - Status: Beendet


- save all changes you have done so far
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}


- 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])
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
- DRAM this build to Ohm (from C:/sources/builder/src/13r1/ip6010/bin) and test the provider profile (V12 AND V13!)
: {{Template:SIP_Profile_Test_T38_PSTN_yes_MR_Ex}}
- 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)
; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A


- checkout/edit Makefile in $/test/13r1/relay/sip-profiles/
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
  - 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
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}


- create the service release documentation
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}
  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
; Mobility Calls {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}
  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 ------------------------------
; SRTP : {{Template:SIP_Profile_Test_SRTP_yes}}
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):  
; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}


  - <explain issues here, mention traces if any>
; Call Transfer :
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consconn}}
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consext}}


We have created a special configuration form four your product (screenshot ProviderProfile.png attached).
Would you please have a look at it and especially review and check the terms we are using in the form?
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.


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


*** Do not forget to attach ProviderProfile.png as well es all relevant traces.
==Configuration==
Use profile ''NL-oneCentral-SIP_Trunk_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
 
Please note the following configuration hints:
* <nowiki>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</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>
 
: {{SIP_TEST_V13_HINT}}
 
== Disclaimer ==
{{SIP_TEST_PREFACE}}
 
[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Revision as of 16:38, 30 March 2020

Summary

Tests for the SIP_Trunk_TLS SIP trunk product of the provider oneCentral were completed. Test results have been last updated on March 30th, 2020. Check the history of this article for the date of the first publication of the testreport.

Remarks

- 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

- Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

<internal>Provider SBC: Hermes SNS</internal>


List of Issues found in media-relay Configuration

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.
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.
RALERT DISC
Call disconnected by far end during alert does not disconnect locally
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.
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.

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_VIDEO, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, SUBSCRIBER_NR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS


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 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 supported by the provider. The provider has a failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for a certain duration (i.e default SIP-registration interval) incoming and outgoing calls might be rejected/fail.|timeout=2 minutes}}
Correct signalling of Ringing-state
OK
An outgoing call that is disconnected by the far end during alert is not disconnected locally. Typically, the provider may play a message to the effect that the call has been rejected or the call can not be completed so that the calling user will hang up. This may be OK, however, it may create issues with automated calls (e.g. fax or modem) which do not listen to the announcement and keep waiting for the far end to accept the call.
CLIR
OK
Clip No Screening (CLNS)
CLIP no screening (CLNS) is not possible, that is, outgoing calls can only have the own subscriber number (or extension), no foreign numbers as calling line id. This affects for example externally forwarded and mobility calls.
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 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
Transport of faxes to/from the PSTN via G.711 codec was tested successfully.
Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. However since the provider requires the MediaRelay and Exclusive-Coder setting, T.38 is not activated on the SIP-interface. The reason for this is the non-working fallback to AudioFax of the Fax-interface, in case that the SIP-interface is configured with above options. This limitation applies only to the Fax-interface. If you are not using it for fax calls, you can enable T.38 by using the "Expert Mode" at the SIP-profile.
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.
SRTP
The provider supports audio encryption using SRTP for incoming, outgoing and on-net calls.
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
The provider does not handle internally transferred-after-connect calls.
The provider does not handle externally transferred calls.


Configuration

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

Please note the following configuration hints:

  • 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
  • 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.