Howto:FR - Peoplefone - SIP trunk TCP SIP-Provider (2024)

Summary

Tests for the SIP_trunk_TCP SIP trunk product of the provider Peoplefone were completed. Test results have been last updated on July 2nd, 2024. Check the history of this article for the date of the first publication of the testreport.

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

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

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 and TCP as transport protocol. The tests were completed using TCP, since UDP is an unreliable protocol and 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 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

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

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

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

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

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

OK

Call Transfer

OK

Configuration

Use profile FR-Peoplefone-SIP_trunk_TCP 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

A most recent v14r1 firmware is required to use this SIP-profile. For hints regarding upgrade to v14r1, see Howto:V14_Firmware_Upgrade_V13r3_V14r1

New profiles are added in the course of our V14R1 software Service Releases, see ReleaseNotes14r1: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.