Howto:DE - CITEC AG - Universe SIP Trunk SIP-Provider (2017)

Summary

Tests for the [https: Universe_SIP_Trunk] SIP trunk product of the provider CITEC_AG were completed. Test results have been last updated on December 28th, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

• CLNS only works if CGPN number sent has 12 digits (the 00 does not count), if shorter the call will be displayed as anonymous

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

• 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects

• Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

<internal>Provider SBC: User-Agent: Universe 1.0</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.

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 302

The provider does not support external call redirection using the SIP 302 Redirect response

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.

REVERSE MEDIA

The provider does not support reverse media negotiation (a.k.a. late SDP)

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, G722, HOLD_RETRIEVE, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, MOBILITY, 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, G729, G729_ONNET, OPUS_WB, OPUS_NB, CLNS, CLNS_ONNET

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 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. 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 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.

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 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 G722

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

OK

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 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

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.

Configuration

Use profile DE-CITEC_AG-Universe_SIP_Trunk 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
• Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

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 ReleaseNotes12r2: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.