Howto:ES - Voztelecom - SIP Trunk SIP-Provider (2021)

Summary

Tests for the SIP_Trunk SIP trunk product of the provider Voztelecom were completed. Test results have been last updated on April 30th, 2021. 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.

CLNS ONNET

Onnet-Calls (that is, within the provider's network) do not allow foreign calling party numbers (CGPN). In other words, clip no screening is not possible for on-net calls.

CLNS

Outgoing calls cannot be sent with a foreign calling party number (CLI).

FAX AUDIO

The provider does not fully support Audiofax (i.e. non-T.38)

FAX T38 ONNET

The provider does not support T.38 fax for onnet calls.

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.

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

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.

REVERSE MEDIA

The provider does not support reverse media negotiation (a.k.a. late SDP)

SDP ICE

The provider does not support receiving ICE candidates in the SDP-part of a SIP message.

SDP RTCP MUX

The provider does not support receiving a RTCP-MUX attribute in the SDP-part of a SIP message.

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.

XFER CONS EXT

The provider does not fully support external consultation call transfer scenarios.

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.

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 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 not successful. This will result in unwanted call termination on calls exceeding a certain time (default 30 minutes). Because of this, further tests were aborted.

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.

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)

CLIP no screening (CLNS) is not possible, that is, outgoing calls can only have the own subscriber number (or extension), no foreign numbers as calling line id. This affects for example externally forwarded and mobility calls.

Also, there is no other method available for this provider to make sure externally forwarded calls will show proper calling line identification (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the forwarded call will at least carry the diverting party number (DGPN) as calling party number (CGPN/CLI) when alerting at the forwarded-to target. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.

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

Audio-Fax calls (that is, fax calls without T.38) do not work. However, all fax endpoints must be configured with exclusive codec "G711A".

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, G711U and G729

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

Neither Clip no screening (CLNS) nor call redirection using the SIP Diversion: or History-Info is supported. Calls forwarded to mobility devices will thus not have the original callers calling line id (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the mobility call will at least carry the mobility user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.

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

The provider does not handle externally transferred calls.

Configuration

Use profile ES-Voztelecom-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

• FAX requires exclusive G711A codec
• Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects
• 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects

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.