Howto:DE - autphone - aut trunk SIP-Provider (2021): Difference between revisions

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== Summary ==  
== Summary ==  
{{Template:SIP_TEST_STATUS_ongoing|update=February 15th, 2021|url=|productname=Autphone|providername=DE}}
{{Template:SIP_TEST_STATUS_ongoing|update=March 1st, 2021|url=https://autphone.com/en/product/aut-trunk|productname=AutTrunk|providername=Autphone}}
<internal>Provider SBC: </internal>
<internal>Provider SBC: </internal>


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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}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; 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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{{SIP_TEST_ISSUES_ALTERNATE_INTRO}}
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
{{SIP_TEST_ISSUES_NO_ALTERNATE_ISSUES}}
; MOBILITY : {{SIP_TEST_FACT_WORKSINALTERNATE_NOT_IN_PRIMARY}}
 


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


; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}
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; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}


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


; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}
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==Configuration==
==Configuration==
Use profile ''DE-DE-Autphone'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
Use profile ''DE-Autphone-AutTrunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.


Please note the following configuration hints:
Please note the following configuration hints:

Revision as of 12:38, 1 March 2021

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated March 1st, 2021) and may (and probably will) change. <internal>Provider SBC: </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.

EARLY MEDIA INBOUND
The provider does not support early-media (i.e. establish RTP-stream before 200 OK/connect) for calls to the PSTN.
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.
MOBILITY
The provider can not send DTMF signals via SIP-INFO messages.
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: 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.

MOBILITY
This feature, which does not work in the first configuration, works fine in the second configuration.


Test Results

This section explains the test results for all possible configurations in more detail.

Configuration without media-relay

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.
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.
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}}
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.
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. This provider supports call redirection using the SIP 302 Redirect header. 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, this configuration is not strictly required, as the provider supports clip no screening so that redirected calls (i.e. call forwards to external numbers) will show a proper calling line id (CLI) at the receiving party anyway. However, it may be useful anyhow to get rid of externally forwarded calls on the SBC entirely.
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. However, all fax endpoints must be configured with exclusive codec "G711A".
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 and G711U
supported onnet (VoIP to VoIP): G711A, G711U and G722
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. In a no-media-relay configuration, DTMF signalling can thus not be conveyed to the PBX. Mobility calls will not work.
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 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.
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.
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}}
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.
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. This provider supports call redirection using the SIP 302 Redirect header. 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, this configuration is not strictly required, as the provider supports clip no screening so that redirected calls (i.e. call forwards to external numbers) will show a proper calling line id (CLI) at the receiving party anyway. However, it may be useful anyhow to get rid of externally forwarded calls on the SBC entirely.
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. However, all fax endpoints must be configured with exclusive codec "G711A".
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 and G711U
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
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 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 DE-Autphone-AutTrunk in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

  • FAX requires exclusive G711A codec
  • 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.