Howto:DE - outbox AG - Whitebox SIP SIP-Provider (2020): Difference between revisions
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; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}} | ; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}} | ||
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}} | ; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}} | ||
; CLNS : {{SIP_TEST_FACT_CLNS}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
; DTMF : {{SIP_TEST_FACT_DTMF}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}} | ; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}} | ||
; FAX T38 : {{SIP_TEST_FACT_FAX T38}} | ; FAX T38 : {{SIP_TEST_FACT_FAX T38}} | ||
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}} | ; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}} | ||
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}} | |||
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}} | ; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}} | ||
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}} | ; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}} | ||
; SDP ICE : {{SIP_TEST_FACT_SDP ICE}} | ; SDP ICE : {{SIP_TEST_FACT_SDP ICE}} | ||
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}} | ; SIP INFO : {{SIP_TEST_FACT_SIP INFO}} | ||
; XFER CONS EXT : {{SIP_TEST_FACT_XFER CONS EXT}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
; XFER CONS : {{SIP_TEST_FACT_XFER CONS}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
<small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]</small> | <small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]</small> | ||
Line 23: | Line 30: | ||
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}} | {{SIP_TEST_ISSUES_ALTERNATE_INTRO}} | ||
{{SIP_TEST_ISSUES_MR_INTRO}} | {{SIP_TEST_ISSUES_MR_INTRO}} | ||
; | ; CLNS : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}} | ||
; | ; DTMF : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}} | ||
; | ; MOBILITY : {{SIP_TEST_FACT_WORKSINALTERNATE_NOT_IN_PRIMARY}} | ||
: {{ | ; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}} | ||
; | ; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}} | ||
; XFER CONS : {{ | ; XFER CONS EXT : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}} | ||
: {{ | ; XFER CONS : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}} | ||
Line 39: | Line 46: | ||
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}} | ; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}} | ||
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}} | ; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}} {{Template:SIP_Profile_Test_DTMF_unreliable}} | ||
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}} | ; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}} | ||
Line 49: | Line 56: | ||
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}} | ; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}} | ||
; Clip No Screening (CLNS) : {{Template: | ; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_history_or_diversion}} | ||
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ||
Line 58: | Line 65: | ||
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}} | : {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}} | ||
; Codecs : supported to/from PSTN: G711A | ; Codecs : supported to/from PSTN: G711A and G711U | ||
: supported onnet (VoIP to VoIP): G711A and G711U | : supported onnet (VoIP to VoIP): G711A and G711U | ||
Line 73: | Line 80: | ||
; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}} | ; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}} | ||
; Call Transfer : {{Template: | ; Call Transfer : | ||
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consconn}} | |||
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consext}} | |||
Line 100: | Line 109: | ||
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}} | : {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}} | ||
; Codecs : supported to/from PSTN: G711A | ; Codecs : supported to/from PSTN: G711A | ||
: supported onnet (VoIP to VoIP): G711A and G711U | : supported onnet (VoIP to VoIP): G711A and G711U | ||
Line 115: | Line 124: | ||
; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}} | ; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}} | ||
; Call Transfer | ; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}} | ||
: {{Template: | |||
Revision as of 11:17, 7 April 2020
Summary
Tests for the Whitebox_SIP SIP trunk product of the provider outbox_AG were completed. Test results have been last updated on April 7th, 2020. 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 no media-relay Configuration
This is a list of all issues found in a configuration where the media stream between endpoints and the SIP provider - as opposed to the signalling - is not routed through the SBC.
- 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).
- 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.
- 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 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.
- 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 - 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.
- 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.
- XFER CONS
- The provider does not fully support consultation call transfer after connect 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
List of Issues found in media-relay Configuration
This section lists the results that differ from the results for the first configuration.
- CLNS
- This feature, which is unstable in the first configuration, works fine in the second configuration.
- DTMF
- This feature, which is unstable in the first configuration, works fine in the second configuration.
- MOBILITY
- This feature, which does not work in the first configuration, works fine in the second configuration.
- 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. - XFER CONS EXT
- This feature, which is unstable in the first configuration, works fine in the second configuration.
- XFER CONS
- This feature, which is unstable in the first configuration, works fine in the second configuration.
Test Results
This section explains the test results for all possible configurations in more detail.
Configuration without 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. 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 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
- 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.
- 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.
- 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 and G711U
- supported onnet (VoIP to VoIP): G711A and G711U
- 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.
- Dialing of Subscriber Numbers
- OK
- Call Transfer
- The provider does not handle internally transferred-after-connect calls.
- The provider does not handle externally transferred calls.
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 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 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
- 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:
orDiversion:
SIP headers for providing the correct calling line id for diverted calls. However, on-net (that is, from SIP provider to another customer at the same SIP provider) CLIP no screening (CLNS) is not possible.
- 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.
- 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 and G711U
- 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.
- Dialing of Subscriber Numbers
- OK
- Call Transfer
- OK
Configuration
Use profile DE-outbox_AG-Whitebox_SIP 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 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.