Howto:NL - infopact - SIP TRUNK SIP-Provider (2019): Difference between revisions
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== Summary == | == Summary == | ||
{{Template: | {{Template:SIP_TEST_STATUS_complete|update=May 13th, 2019|url=https://infopact.nl/telefonie/sip-trunk/|productname=SIP_Trunk|providername=infopact}} | ||
<internal>Provider SBC: Infopact</internal> | <internal>Provider SBC: Infopact</internal> | ||
=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} === | === {{SIP_TEST_ISSUES_NO_MR_TITLE}} === | ||
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; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}} | ; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}} | ||
; CLNS : {{SIP_TEST_FACT_CLNS}} | ; CLNS : {{SIP_TEST_FACT_CLNS}} | ||
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
; 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 25: | Line 25: | ||
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}} | {{SIP_TEST_ISSUES_ALTERNATE_INTRO}} | ||
{{SIP_TEST_ISSUES_MR_INTRO}} | {{SIP_TEST_ISSUES_MR_INTRO}} | ||
; MOBILITY : {{ | ; 180 RINGING : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}} | ||
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
; RALERT DISC : {{SIP_TEST_FACT_RALERT DISC}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
== Test Results == | == Test Results == | ||
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; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ||
; Early-Media : {{Template: | ; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}} | ||
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ||
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; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}} | ; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}} | ||
; Mobility Calls : {{Template: | ; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_without_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}} | ||
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}} | ; SRTP : {{Template:SIP_Profile_Test_SRTP_no}} | ||
; Dialing of Subscriber Numbers : {{Template: | ; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}} | ||
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}} | ; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}} | ||
Line 82: | Line 87: | ||
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}} | ; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}} | ||
; Correct signalling of Ringing-state : {{Template: | ; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}} | ||
:{{Template:SIP_Profile_Test_RALERT_DISC_no}} | |||
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}} | ; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}} | ||
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; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ||
; Early-Media : {{Template: | ; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}} | ||
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ||
Line 104: | Line 110: | ||
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}} | ; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}} | ||
; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay | ; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} | ||
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}} | ; SRTP : {{Template:SIP_Profile_Test_SRTP_no}} | ||
; Dialing of Subscriber Numbers : {{Template: | ; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}} | ||
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}} | ; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}} | ||
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Please note the following configuration hints: | Please note the following configuration hints: | ||
* <nowiki>If you want to use the Mobility feature, please use only the Media Relay configuration!</nowiki> | |||
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki> | * <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki> | ||
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki> | |||
: {{SIP_TEST_V12_HINT}} | : {{SIP_TEST_V12_HINT}} |
Revision as of 21:58, 13 May 2019
Summary
Tests for the SIP_Trunk SIP trunk product of the provider infopact were completed. Test results have been last updated on May 13th, 2019. Check the history of this article for the date of the first publication of the testreport. <internal>Provider SBC: Infopact</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. - 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.
- CLNS
- Outgoing calls cannot be sent with a foreign calling party number (CLI).
- 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 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 - 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.
- 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, 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.
- 180 RINGING
- This feature, which is unstable in the first configuration, works fine in the second configuration.
- 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.
Test Results
This section explains the test results for all possible configurations in more detail.
Configuration without media-relay
- Registration
- The provider supports UDP and TCP as transport protocol. The tests were completed using TCP, since UDP is an unreliable protocol and 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 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 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.
- 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
- 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.
- 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.
- 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
Configuration with media-relay
- Registration
- The provider supports UDP and TCP as transport protocol. The tests were completed using TCP, since UDP is an unreliable protocol and 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 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
- 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.
- 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
- IP-Fragmentation
- OK
- Large SIP messages
- OK
- Reverse Media Negotiation
- OK
- Mobility Calls
- Transmitting DTMF-tones as SIP-INFO messages is not supported, however mobility calls are still possible as in the media-relay configuration, the SBC will convey DTMF signalling to the PBX.
- SRTP
- The provider does not support audio encryption using SRTP.
- 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
Configuration
Use profile NL-infopact-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.
Please note the following configuration hints:
- If you want to use the Mobility feature, please use only the Media Relay configuration!
- Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects
- Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
- 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.