Howto:DE - Colt - SIP Trunking SIP-Provider (2017)

Summary

Tests for the SIP_Trunking SIP trunk product of the provider Colt were completed. Test results have been last updated on December 21st, 2017. 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

180 RINGING

The provider does not send a 180 Ringing response when the called party alerts.

RALERT DISC

Call disconnected by far end during alert does not disconnect locally

REDIR 302

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

SDP VIDEO

The provider does not support receiving video media capabilities in the SDP-part of a SIP message.

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, LARGE_SIP_MESSAGES, MOBILITY, OPUS_NB, OPUS_WB, RALERT_DISC, REDIR_302, IP_FRAGMENTATION, 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.

Summary

Tests for the SIP_Trunking SIP trunk product of the provider Colt were completed. Test results have been last updated on December 21st, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

• encryption using SRTP is also offered on request by the provider, however not tested during this interop tests
• using TCP as signaling protocol is also offered on request by the provider, however not tested during this interop tests
• the provider offers SIP Trunking in various other countries. Please contact the provider to check if the tested German account has the same specifications as the account in the country you are interested in.

<internal>Provider SBC: </internal>

List of Issues found in media-relay Configuration

180 RINGING

The provider does not send a 180 Ringing response when the called party alerts.

RALERT DISC

Call disconnected by far end during alert does not disconnect locally

REDIR 302

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

SDP VIDEO

The provider does not support receiving video media capabilities in the SDP-part of a SIP message.

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, LARGE_SIP_MESSAGES, MOBILITY, OPUS_NB, OPUS_WB, RALERT_DISC, REDIR_302, IP_FRAGMENTATION, 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 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

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

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.

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)

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.

Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.

Codecs

supported to/from PSTN: G711A and G729

supported onnet (VoIP to VoIP): G711A and G729

Large SIP messages

OK

IP-Fragmentation

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 DE-Colt-SIP_Trunking 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 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.