Howto:AT - yuutel - SIP Trunk TLS SIP-Provider (2025)

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated April 15th, 2025) and may (and probably will) change.

List of Issues found in media-relay Configuration

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.

CLNS

Outgoing calls cannot be sent with a foreign calling party number (CLI).

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 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

FAX T38ANDAUDIO

The provider does not support fallback to audio-fax if T.38 fails.

LARGE SIP MESSAGES

The provider does not support large SIP messages (> 1500 bytes).

MOBILITY

The provider can not send DTMF signals via SIP-INFO messages.

RALERT DISC

Call disconnected by far end during alert does not disconnect locally

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)

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

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, 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 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

An outgoing call that is disconnected by the far end during alert is not disconnected locally. Typically, the provider may play a message to the effect that the call has been rejected or the call can not be completed so that the calling user will hang up. This may be OK, however, it may create issues with automated calls (e.g. fax or modem) which do not listen to the announcement and keep waiting for the far end to accept the call.

CLIR

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. Call redirection using the SIP 302 Redirect header is supported by the provider. 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.

Also, there is no other method available for this provider to make sure externally forwarded calls will show proper calling line identification (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the forwarded call will at least carry the diverting party number (DGPN) as calling party number (CGPN/CLI) when alerting at the forwarded-to target. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.

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

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

supported onnet (VoIP to VoIP): G711A, G711U and G722

IP-Fragmentation

OK

Large SIP messages

Large SIP messages (> 1500 bytes) are not supported by the provider. This might lead to sporadic failure of outbound calls, e.g. if the call has redirection information and by additional data the singling message gets to large for the SIP-provider.

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.

Neither Clip no screening (CLNS) nor call redirection using the SIP Diversion: or History-Info is supported. Calls forwarded to mobility devices will thus not have the original callers calling line id (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the mobility call will at least carry the mobility user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.

SRTP

The provider supports audio encryption using SRTP for onnet calls.

Dialing of Subscriber Numbers

OK

Call Transfer

OK

Configuration

Use profile AT-yuutel-SIP_Trunk_TLS in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

• 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects
• If you intend to use SIPS (SIP/TLS) registration, you need to add the ' p02.neotel.at' certificate to the trust list of your SBC

A most recent v15r1 firmware is required to use this SIP-profile. For hints regarding upgrade to v15r1, see Howto15r1:Firmware_Upgrade_V14r2_V15r1

New profiles are added in the course of our V15R1 software Service Releases, see ReleaseNotes15r1: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.