Howto:DE - Swissnet - SIP Trunk SIP-Provider (2023)

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Contents 1 Summary 1.1 List of Issues found in media-relay Configuration 2 Test Results 2.1 Configuration with media-relay 3 Configuration 4 Disclaimer

[edit] Summary

Tests for the [ SIP_Trunk] SIP trunk product of the provider Swissnet were completed. Test results have been last updated on March 10th, 2023. Check the history of this article for the date of the first publication of the testreport.

[edit] List of Issues found in media-relay Configuration

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 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.

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

[edit] 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.

[edit] 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.

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. 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}}

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.

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 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 G729

supported onnet (VoIP to VoIP): G711A and G729

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 does not support audio encryption using SRTP. Encryption is done asymmetrical, i.e. either calls from or to the PSTN are not encrypted.

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

[edit] Configuration

Use profile DE-Swissnet-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

• 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.

[edit] 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.