Howto:ES - LCRcom - Voz IP SIP-Provider (2017): Difference between revisions

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== Summary ==  
== Summary ==  
{{Template:SIP_TEST_STATUS_complete|update=September 13th, 2017|url=https://www.lcrcom.net/empresas/soluciones-empresas/voz-ip|productname=Voz_IP|providername=LCRcom}}
{{Template:SIP_TEST_STATUS_complete|update=October 9th, 2017|url=https://www.lcrcom.net/empresas/soluciones-empresas/voz-ip|productname=Voz_IP|providername=LCRcom}}
=== Remarks ===
=== Remarks ===
* SRTP can be activated by the provider. This template tested the default settings for the product Voz IP with innovaphone Gateways.  
* SRTP can be activated by the provider. This template tested the default settings for the product Voz IP with innovaphone Gateways.  
* T38 can be activated as well by contacting the provider. It was not part of this test.
* T38 can be activated as well by contacting the provider. It was not part of this test.  
* CLNS can be activated by the provider


<internal>Provider SBC: ENSR3.0.67.5-IS4-RMRG1656-RG803-CPI1-CPO6271</internal>
<internal>Provider SBC: ENSR3.0.67.5-IS4-RMRG1656-RG803-CPI1-CPO6271</internal>
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=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
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; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; RALERT DISC : {{SIP_TEST_FACT_RALERT DISC}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}


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; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}


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


; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}


; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_no_clns_302_recommended}}
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_no_clns_302_not_recommended}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}


; 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}}
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; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}


; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}


; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_clns_no_history_or_diversion}}
; 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}}
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
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Please note the following configuration hints:
Please note the following configuration hints:
* 'Reroute supported' recommended in PBX 'Trunk' objects
* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects
* 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects


: {{SIP_TEST_V12_HINT}}
: {{SIP_TEST_V12_HINT}}

Revision as of 17:40, 9 October 2017

Summary

Tests for the Voz_IP SIP trunk product of the provider LCRcom were completed. Test results have been last updated on October 9th, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

  • SRTP can be activated by the provider. This template tested the default settings for the product Voz IP with innovaphone Gateways.
  • T38 can be activated as well by contacting the provider. It was not part of this test.
  • CLNS can be activated by the provider

<internal>Provider SBC: ENSR3.0.67.5-IS4-RMRG1656-RG803-CPI1-CPO6271</internal>


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.
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.
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.
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_EXT, XFER_CONS_ALERT, 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 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. 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)
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. Call redirection using the SIP 302 Redirect header is supported by the provider. Rerouting can be enabled by setting the Reroute supported check-mark in the corresponding Trunk object configuration. Also, the Interworking check-mark must be set in the route used for calls from and to the SIP provider.
However, during our test other interop problems were discovered when the Interworking Flag is enabled. Therefore it is not recommended to use the call redirection via SIP 302 Redirect header.
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 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, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U and G729
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.
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.
Dialing of Subscriber Numbers
OK
Call Transfer
OK


Configuration

Use profile ES-LCRcom-Voz_IP 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
  • 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects
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.