Howto:DE - htp - SIP Trunk pure SIP-Provider (2026)

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated April 1st, 2026) and may (and probably will) change.

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.

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.

SDP RTCP MUX

The provider does not support receiving a RTCP-MUX attribute in the SDP-part of a SIP message.

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.

SIP INFO

The provider does not support conveying DTMF using the SIP-INFO method.

XFER CONS ALERT

The provider does not fully support consultation call transfer after alert scenarios.

XFER CONS

The provider does not fully support consultation call transfer after connect scenarios.

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.

The test results for this configuration are the same, however.

Test Results

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

Configuration without media-relay

Registration

The provider supports UDP, TCP and TLS as transport protocol. The tests were completed using TLS, since it offers encryption of the transmitted SIP-packets.

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

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, during our test other interop problems were discovered when the Interworking Flag is enabled. Therefore it is not recommended to use the call redirection via SIP 302 Redirect header.

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

supported onnet (VoIP to VoIP): G711A and G711U

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 supports audio encryption using SRTP for incoming, outgoing and on-net calls.

Dialing of Subscriber Numbers

OK

Call Transfer

The provider does not handle internally transferred-after-connect calls.

The provider does not handle internally transferred-after-alert calls.

Configuration with media-relay

Registration

The provider supports UDP, TCP and TLS as transport protocol. The tests were completed using TLS, since it offers encryption of the transmitted SIP-packets.

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

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, during our test other interop problems were discovered when the Interworking Flag is enabled. Therefore it is not recommended to use the call redirection via SIP 302 Redirect header.

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

supported onnet (VoIP to VoIP): G711A and G711U

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 supports audio encryption using SRTP for incoming, outgoing and on-net calls.

Dialing of Subscriber Numbers

OK

Call Transfer

The provider does not handle internally transferred-after-connect calls.

The provider does not handle internally transferred-after-alert calls.

Configuration

Use profile DE-htp-SIP_Trunk_pure in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

• FAX requires exclusive G711A codec
• STUN required on all endpoints if media-relay is not used
• If you intend to use SIPS (SIP/TLS) registration, you need to add the ' DigiCert Global Root G2' certificate to the trust list of your SBC

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.