Howto:IT - Made in Lab - VoIP In Lab SIP-Provider (2016): Difference between revisions

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== Summary ==
== Summary ==  
<!--{{Template:Compat Status "planned"}} -->
The tests for the [http://www.m-lab.it/en/ ''VoIP-In-Lab]'' SIP trunk product of the provider ''Made-in-Lab'' were completed.
<!-- {{Template:Compat Status "in progress"}} -->
{{Template:Compat Status "tested"(sip provider)}}
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


The tests for the SIP-trunk ''VoIP-In-Lab'' of the provider [http://www.m-lab.it MADE IN LAB Communications] were completed.
Testing of this product has been finalized August 10th, 2016.


Issues found were:
; SRTP
; Reverse Media Negotiation
; Mobility Call
; Redundancy
; Correct signalling of Ringing-state


For more details about this issues, see ''Tests'' sections.


== Current test state ==
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
<!--{{Template:Compat Status "planned"}} -->
{{SIP_TEST_ISSUES_MR_INTRO}}
<!-- {{Template:Compat Status "in progress"}} -->
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
{{Template:Compat Status "tested"(sip provider)}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
<!-- {{Template:Compat Status "rejected"}} -->
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}


Testing of this product has been finalized February 11th, 2016.
<small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[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_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[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_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_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[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_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[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_SIP_INFO_FAILS|SIP_INFO]]</small>


[[Category:RecProd|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]


== Tests with MediaRelay ==
== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}
 
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}
 
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}
 
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}
 
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_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}}
 
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}
 
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
 
; Codecs : supported to/from PSTN: G711A, G711U and G729
: supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
 
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}
 
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_yes}}
 
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
 
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}
 
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
 
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}
 
; Mobility Call :  {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}


; Registration : The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted. Moreover it requires all involved network elements to support IP-fragmentation.
; CLIP : OK
; CLIR : OK
; CLNS : OK
; codecs : ''G711a, G711u, G729A, G722 (on-net calls only)''. The following codecs are not supported: ''OPUS''.
; Fax : Transport of faxes to/from the PSTN via G.711A codec was tested successfully.Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.
; SRTP : The provider does not support audio encryption using SRTP.
; DTMF (RFC2833) : OK
; 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.
; Reverse Media Negotiation : Reverse-media negotiation is not supported, MediaRelay and Exclusive coder must be activated on the SIP-interface.
; Mobility Call : Transmitting DTMF-tones as SIP-INFO messages was not possible, which is required in-order to use Mobility-calls without MediaRelay on the SIP-Interface.
; Redundancy : Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no 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 duration of up to 2 minutes(i.e registration interval) incoming and outgoing calls might be rejected/fail.
; IP-Fragmentation : OK
; Large SIP messages : OK
; Correct signalling of Ringing-state : Ringing 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, will get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call.
; Early-Media : OK
; Session Timer : OK
; Call Transfer : OK


== Tests without MediaRelay ==
The tests were aborted, since the provider requires MediaRelay. The reason for it, are audio problems when changing the remote RTP-endpoint during a call and missing support for reverse-media negotiation.


==Configuration==
==Configuration==
Use profile ''IT-Made-in-Lab-VoIP-In-Lab'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.


* Use profile ''IT-Made_in_Lab-VoIP_In_Lab'' in the Gateway/Interfaces/SIP menu.
== Disclaimer ==
{{SIP_TEST_PREFACE}}


==Contact==
[[Category:Compat|{{PAGENAME}}]]
Customers should contact ''Made in Lab'' using the following [http://m-lab.it/contatti-2/ contact web page].
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Revision as of 14:07, 10 August 2016

Summary

The tests for the VoIP-In-Lab SIP trunk product of the provider Made-in-Lab were completed.

Testing of this product has been finalized August 10th, 2016.


List of Issues found in media-relay Configuration

180 RINGING
The provider does not send a 180 Ringing response when the called party alerts.
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: BASIC_CALL, 180_RINGING, CLIR, CLNS, CONN_NR, DTMF, EARLY_MEDIA_INBOUND, G711A, G711A_ONNET, G711U, G711U_ONNET, G722, G722_ONNET, G729, G729_ONNET, HOLD_RETRIEVE, MOBILITY, OPUS_NB, OPUS_WB, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, XFER_CONS_EXT, XFER_CONS, XFER_CONS_ALERT, XFER_BLIND, FAX_T38, FAX_AUDIO, SRTP_OUTGOING, SRTP_INCOMING, SRTP_INTERNAL, REVERSE_MEDIA, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, SIP_INFO


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

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.
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.
Registration
The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted(TLS). Moreover it requires all involved network elements to support IP-fragmentation.
Redundancy
Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no 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 2 minutes incoming and outgoing calls might be rejected/fail.
DTMF (RFC2833)
The provider can convey DTMF digits using the RTP payload method as per RFC2833.
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.
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)
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.
Early-Media
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
Codecs
supported to/from PSTN: G711A, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
Call Transfer
OK
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. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.
SRTP
The provider does not support audio encryption using SRTP.
Reverse Media Negotiation
Reverse-media negotiation is not supported. Therefore, a media-relay with exclusive coder configuration will be activated on the SIP-interface.
IP-Fragmentation
OK
Large SIP messages
OK
Mobility Call
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.


Configuration

Use profile IT-Made-in-Lab-VoIP-In-Lab in Gateway/Interfaces/SIP to configure this SIP provider.

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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