Howto:AT - RTM businessit GmbH - SIP Trunk SIP-Provider (2020)

Summary

Tests for the SIP Trunk SIP trunk product of the provider RTM businessit GmbH were completed. Test results have been last updated on September 21st, 2020. Check the history of this article for the date of the first publication of the testreport. <internal>Provider SBC: </internal>

List of Issues found in media-relay Configuration

EARLY MEDIA INBOUND

The provider does not support early-media (i.e. establish RTP-stream before 200 OK/connect) for calls to the PSTN.

REDIR 302

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

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: CLIR, CLNS_ONNET, CLNS, CONN_NR_INCOMING, CONN_NR, DTMF, FAX_AUDIO, FAX_T38_ONNET, FAX_T38, FAX_T38ANDAUDIO, 180_RINGING, BASIC_CALL, CONN_NR_DIFF, EARLY_MEDIA_INBOUND, 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

Signalling protocol

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 does not support conveying telephony events (a.k.a. DTMF) in-band as RTP payload.

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

Currently we don't test the redundancy for SIP-trunks without registration.

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.

Fax

Transport of faxes to/from the PSTN via G.711 codec was tested successfully.

Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. However since the provider requires the MediaRelay and Exclusive-Coder setting, T.38 is not activated on the SIP-interface. The reason for this is the non-working fallback to AudioFax of the Fax-interface, in case that the SIP-interface is configured with above options. This limitation applies only to the Fax-interface. If you are not using it for fax calls, you can enable T.38 by using the "Expert Mode" at the SIP-profile.

Correct signalling of Ringing-state

OK

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).

Codecs

supported to/from PSTN: G711A

supported onnet (VoIP to VoIP): G711A

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.

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

Configuration

Use profile AT-RTM_businessit_GmbH-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'
• This is a SIP account without registration. Depending on the provider a firewall rule is necessary.

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.

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.