Howto:LU - MIXvoip - SIP Trunk SIP-Provider (2021): Difference between revisions
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== Summary == | == Summary == | ||
The tests for the [ | The tests for the ''[http://www.innovaphone.com/redirect.php?url=https%3A%2F%2Fwww.mixvoip.com%2Fsip_trunking%2F // bla SIP_Trunk]'' SIP trunk product of the provider ''MIXvoip'' were completed. | ||
Test results for this product have been last updated August 16th, 2016. | |||
=== Remarks === | |||
We are still investigating the fax issue | |||
== | === {{SIP_TEST_ISSUES_MR_TITLE}} === | ||
{{SIP_TEST_ISSUES_MR_INTRO}} | |||
; CLIR : {{SIP_TEST_FACT_CLIR}} | |||
; CLNS : {{SIP_TEST_FACT_CLNS}} | |||
; DTMF : {{SIP_TEST_FACT_DTMF}} | |||
: {{SIP_TEST_FACT__unreliable}} | |||
; FAX T38 : {{SIP_TEST_FACT_FAX T38}} | |||
< | <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_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_FAILS|FAX_T38]], [[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_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_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> | ||
== | == Test Results == | ||
; Registration : | {{SIP_TEST_TESTRESULT_ONLYMR_INTRO}} | ||
=== {{SIP_TEST_RESULTS_MR_TITLE}} === | |||
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP}} | |||
; | ; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}} | ||
; | ; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_no}} {{Template:SIP_Profile_Test_DTMF_RFC2833_no_but_INFO}} {{Template:SIP_Profile_Test_DTMF_unreliable}} | ||
; | ; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}} | ||
: | |||
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}} | |||
; | ; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}} | ||
; | ; CLIR : {{Template:SIP_Profile_Test_CLIR_no}} | ||
; | ; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_no_clns_but_clns_onnet_OK}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_history_or_diversion}} | ||
; | ; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}} | ||
; | ; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="unknown"}} | ||
: {{Template:SIP_Profile_Test_T38_PSTN_no}} | |||
; | ; Codecs : supported to/from PSTN: G711A | ||
: supported onnet (VoIP to VoIP): G711A | |||
; | ; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}} | ||
; | ; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}} | ||
; | ; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}} | ||
; | ; Mobility Call : {{Template:SIP_Profile_Test_MobilityCall_yes}} | ||
; | ; SRTP : {{Template:SIP_Profile_Test_SRTP_no}} | ||
; | ; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}} | ||
==Configuration== | ==Configuration== | ||
Use profile ''LU-MIXvoip-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider. | |||
== Disclaimer == | |||
{{SIP_TEST_PREFACE}} | |||
[[Category:Compat|{{PAGENAME}}]] | |||
[[Category:3rdParty SIP Provider|{{PAGENAME}}]] |
Revision as of 13:15, 16 August 2016
Summary
The tests for the // bla SIP_Trunk SIP trunk product of the provider MIXvoip were completed.
Test results for this product have been last updated August 16th, 2016.
Remarks
We are still investigating the fax issue
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
- Outgoing calls cannot be sent with a foreign calling party number (CLI).
- 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
- The provider does not fully support T.38 fax
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, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38, 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, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, 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. However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call.
- DTMF (RFC2833)
- The provider does not support conveying telephony events (a.k.a. DTMF) in-band as RTP payload. However, it supports conveying telephony events (DTMF) using the SIP-INFO method. innovaphone endpoints will work with this method too. 3rd party endpoints may have issues though. 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
- 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. However, on-net (that is, from SIP provider to another customer at the same SIP provider) CLIP no screening (CLNS) is possible. As this is a non-German provider, the issue with off-net CLNS could be related to the provider or related to the international PSTN peering. Please consult the SIP provider if CLNS will work for you.
- 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. However, all fax endpoints must be configured with exclusive codec "unknown".
- The provider does not support T.38 fax calls to the PSTN. You need to use audio-fax therefore.
- Codecs
- supported to/from PSTN: G711A
- supported onnet (VoIP to VoIP): G711A
- IP-Fragmentation
- OK
- Large SIP messages
- OK
- Reverse Media Negotiation
- OK
- Mobility Call
- OK
- SRTP
- The provider does not support audio encryption using SRTP.
- Call Transfer
- OK
Configuration
Use profile LU-MIXvoip-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.
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.