Howto:DE - NFON - Sip Trunk Flexx SIP-Provider (2024)
Summary
Tests for the Sip_Trunk_Flexx SIP trunk product of the provider NFON were completed. Test results have been last updated on May 7th, 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
- CLIR
- 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.
- CLNS
- Outgoing calls cannot be sent with a foreign calling party number (CLI).
- 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.
- LARGE SIP MESSAGES
- The provider does not support large SIP messages (> 1500 bytes).
- 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 - 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)
- 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 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
- OK
- CLIR
- 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.
- 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 signalled to the caller.
- However, during our test other interop problems were discovered when the Interworking Flag is enabled. As a result, the update of the connected number cannot be signalled.
- 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, G711U and G722
- IP-Fragmentation
- OK
- Large SIP messages
- Large SIP messages (> 1500 bytes) are not supported by the provider. This might lead to sporadic failure of outbound calls, e.g. if the call has redirection information and by additional data the singling message gets to large for the SIP-provider.
- 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.
- 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
- OK
- Call Transfer
- OK
Configuration
Use profile DE-NFON-Sip_Trunk_Flexx in Gateway/Interfaces/SIP to configure this SIP provider.
Please note the following configuration hints:
- 'Set Calling = Diverting No' recommended in PBX 'Trunk' 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 Reference14r1: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.