Howto:DE - ecotel - SIP Trunk 2.0 SIP-Provider (2016): Difference between revisions

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== Summary ==  
== Summary ==  
{{Template:SIP_TEST_STATUS_complete|update=January 26th, 2017|url=https://www.ecotel.de/index.php/sprache/sip-telefonie|productname=SIP Trunk 2.0|providername=ecotel}}
{{Template:SIP_TEST_STATUS_complete|update=August 30th, 2017|url=https://www.ecotel.de/index.php/sprache/sip-telefonie|productname=SIP Trunk 2.0|providername=ecotel}}
=== Remarks ===
=== Remarks ===
* FAX T.38: according to the provider T.38 is supported depending on the remote destination(onnet and offnet), no T.38 termination is done by the provider.  
* FAX T.38: according to the provider T.38 is supported depending on the remote destination(onnet and offnet), no T.38 termination is done by the provider.  
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=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
; CONN NR : {{SIP_TEST_FACT_CONN NR}}
; 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}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; 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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; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP}}
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP}}


; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}  
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}  


; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}
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; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}


; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}}{{Template:SIP_Profile_Test_COLP_diff_no}}
; COLP : {{Template:SIP_Profile_Test_COLP_out_no_in_yes}}{{Template:SIP_Profile_Test_COLP_diff_no}}


; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}
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: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}


; Codecs : supported to/from PSTN: G711A
; Codecs : supported to/from PSTN: G711A, G711U and G729
: supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
: supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729


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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}}
; Mobility Calls :  {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

Revision as of 17:07, 30 August 2017

Summary

Tests for the SIP Trunk 2.0 SIP trunk product of the provider ecotel were completed. Test results have been last updated on August 30th, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

  • FAX T.38: according to the provider T.38 is supported depending on the remote destination(onnet and offnet), no T.38 termination is done by the provider.
  • FAX T38ANDAUDIO: the fallback to audio doesn't work because of a bug in the innovaphone firmware. Disabling T.38 at the SIP-interfaces can be usesd as a work-around.
  • History-Info: The provider sends a History-Info Header for all incoming calls. The innovaphone PBX interprets this as Diverting Party Number, which can cause problems with the conditional Call-Forward feature of the PBX. As a result, you must enable "Discard received diverting No" at your Trunk-Line object.

<internal>Provider SBC: TELES-mgc-360-mod01-build019</internal>


List of Issues found in media-relay Configuration

CONN NR
For outbound calls to the PSTN the provider transmits an incorrect connected number.
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 302
The provider does not support external call redirection using the SIP 302 Redirect response
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, CONN_NR_INCOMING, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38, FAX_T38_ONNET, 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, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, 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 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
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.
COLP
Inbound calls from the PSTN show the correct connected number. However outbound calls to the PSTN do not.
A caller from the PBX to the PSTN, will receive an incorrect connected number that differs from the dialled number. This might lead to the caller cancelling the call.
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 worked to onnet destinations. However it failed to PSTN destination and moreover fallback to audiofax 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, G722 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.
SRTP
The provider does not support audio encryption using SRTP.
Call Transfer
OK


Configuration

Use profile DE-ecotel-SIP_Trunk_2.0 in Gateway/Interfaces/SIP to configure this SIP provider.

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.