Howto:DE - TELE AG - SIP Trunk SIP-Provider (2016)

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Contents

Summary

Tests for the SIP_Trunk SIP trunk product of the provider TELE_AG were completed. Test results have been last updated on June 2nd, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

  • With v12r2 firmware, outbound calls to the provider are rejected with the status message 400 Bad Request. Use 12r1 firmware instead.



List of Issues found in media-relay Configuration

180 RINGING 
The provider does not send a 180 Ringing response when the called party alerts.
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.
CONN NR INCOMING 
Incoming calls from the PSTN don't show a correct connected number to the calling party.
DTMF 
The provider does not fully support reliable transportation of DTMF signals (DTMF tones are treated separately from voice data). There may be different symptoms like no DTMF at all, no DTMF at the beginning of a call, loss of some DTMF digits in a multi-digit DTMF sequence, duplication of DTMF digits or DTMF digits echoed back to the sender
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 T38 ONNET 
The provider does not support T.38 fax for onnet calls.
FAX T38ANDAUDIO 
The provider does not support fallback to audio-fax if T.38 fails.
REDIR 302 
The provider does not support external call redirection using the SIP 302 Redirect response
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, CONN_NR, CONN_NR_INCOMING, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38, FAX_T38_ONNET, 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, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, REVERSE_MEDIA, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, SUBSCRIBER_NR, RALERT_DISC, CLNS_ONNET, CONN_NR_DIFF


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. 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. However, DTMF handling overall does not work reliably.
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 calls to the PSTN show the correct connected number. However incoming calls from the PSTN do not.
A caller from the PSTN will receive an incorrect connected number that differs from the dialled number. This might lead to the caller cancelling the call.
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 failed to PSTN and onnet destinations. Moreover fallback to audio-fax 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, G722 and G729
IP-Fragmentation 
OK
Large SIP messages 
OK
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.
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.
Call Transfer 
OK
Provider supports dialling subscriber numbers 
OK


Configuration

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

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 Support:DVL-Roadmap Firmware 12r2. 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.

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