Howto:CH - VoipTel Internacional SA - SIP Trunk VoipTel SIP-Provider (2023)
Tests for the SIP_Trunk_Voip_Tell SIP trunk product of the provider VoIP_Tel_Internacional were completed. Test results have been last updated on March 31st, 2023. Check the history of this article for the date of the first publication of the testreport.
- For Fax calls to work (outgoing) it's necessary to ask the Provider for a special FAX TRUNK with a Fax number.
- For doing anonymous calls (CLIR) we can use the prefix number 31 (ie. 3100497031730090) when dialing out.
- The receiving of 180 Ringing message depends on the target destination we are calling, sometimes we can receive from the Public Network 180 Ringing or 183 Session Progress.
<internal>Provider SBC: Voip_Tel-Switch</internal>
List of Issues found in media-relay Configuration
- 180 RINGING
- The provider does not send a
180 Ringingresponse when the called party alerts.
- The provider does not fully support suppression of the calling line id (CLIR) using the SIP Privacy: Id header.
- 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.
- 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.
- The provider can not send DTMF signals via SIP-INFO messages.
- 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
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
- 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 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.
- Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for 2 minutes incoming and outgoing calls might be rejected/fail.
- 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 didn't work.
- 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. Call redirection using the SIP 302 Redirect header is supported by the provider. Rerouting can be enabled by setting the Reroute supported check-mark in the corresponding Trunk object configuration. Also, the Interworking check-mark must be set in the route used for calls from and to the SIP provider.
- However, during our test other interop problems were discovered when the Interworking Flag is enabled. Therefore it is not recommended to use the call redirection via SIP 302 Redirect header.
- 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.
- The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
- Audio-Fax calls (that is, fax calls without T.38) do not work.
- 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.
- supported to/from PSTN: G711A, G711U and G729
- supported onnet (VoIP to VoIP): G711A, G711U and G729
- Large SIP messages
- Reverse Media Negotiation
- 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.
- 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
Use profile CH-VoIP_Tel_Internacional-SIP_Trunk_Voip_Tell 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
- Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
- 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.
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.