Howto:NL - X2com - X2VoIP SIP-Provider (2017): Difference between revisions

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== Summary ==  
== Summary ==  
{{Template:SIP_TEST_STATUS_complete|update=September 6th, 2017|url=http://www.x2com.nl/x2voip/|productname=X2VoIP|providername=X2com}}
{{Template:SIP_TEST_STATUS_complete|update=October 20th, 2017|url=http://www.x2com.nl/x2voip/|productname=X2VoIP|providername=X2com}}
=== Remarks ===
=== Remarks ===
* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects
* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects
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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 and G711U
; 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


Line 96: Line 96:


; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}}
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_yes}}


; Codecs : supported to/from PSTN: G711A and G711U
; 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



Revision as of 14:32, 20 October 2017

Summary

Tests for the X2VoIP SIP trunk product of the provider X2com were completed. Test results have been last updated on October 20th, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

  • Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects

<internal>Provider SBC: SpeakUp Gateway</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.
FAX AUDIO
The provider does not fully support Audiofax (i.e. non-T.38)
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
The provider does not fully support T.38 fax
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 T38ANDAUDIO
The provider does not support fallback to audio-fax if T.38 fails.
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.
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
CLNS
Outgoing calls cannot be sent with a foreign calling party number (CLI).
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.
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, 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, G729_ONNET, HOLD_RETRIEVE, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, MOBILITY, RALERT_DISC, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, REVERSE_MEDIA, G722, G729, CLNS, CLNS_ONNET, OPUS_NB, OPUS_WB, 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.

FAX T38
This feature, which is unstable in the first configuration, works fine in the second configuration.
FAX T38ANDAUDIO
This feature, which is unstable in the first configuration, works fine in the second 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 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
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 signalled to the caller.
Early-Media
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
Fax
Audio-Fax calls (that is, fax calls without T.38) do not work.
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
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.
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 supports audio encryption using SRTP for onnet calls.
Dialing of Subscriber Numbers
OK
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
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
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 signalled to the caller.
Early-Media
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
Fax
Audio-Fax calls (that is, fax calls without T.38) do not work.
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
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.
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.
SRTP
The provider supports audio encryption using SRTP for onnet calls.
Dialing of Subscriber Numbers
OK
Call Transfer
OK


Configuration

Use profile NL-X2com-X2VoIP 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
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.