Howto:DE - MK Network - VoiceConnect SIP-Provider (2016): Difference between revisions
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== Summary == | == Summary == | ||
{{Template:SIP_TEST_STATUS_complete|update= | {{Template:SIP_TEST_STATUS_complete|update=September 13th, 2017|url=https://www.mk.de/produkte/moderne-telefonie-loesung/mk-voiceconnect|productname=VoiceConnect|providername=MK_Network}} | ||
<internal>Provider SBC: TELES.iSWITCH</internal> | <internal>Provider SBC: TELES.iSWITCH</internal> | ||
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; SIP INFO : {{SIP_TEST_FACT_SIP INFO}} | ; SIP INFO : {{SIP_TEST_FACT_SIP INFO}} | ||
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}} | ; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}} | ||
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}} | |||
<small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[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_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[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_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_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]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]]</small> | <small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[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_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[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_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_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]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[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_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[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_SDP_VIDEO_FAILS|SDP_VIDEO]]</small> | ||
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; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} | ; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} | ||
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_no}}{{Template:SIP_Profile_Test_COLP_diff_no}} | |||
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}} | ; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}} | ||
| Line 48: | Line 51: | ||
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ||
: {{Template: | : {{Template:SIP_Profile_Test_T38_PSTN_yes_MR_Ex}} | ||
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}} | ; SRTP : {{Template:SIP_Profile_Test_SRTP_no}} | ||
| Line 56: | Line 59: | ||
==Configuration== | ==Configuration== | ||
Use profile ''DE-MK_Network-VoiceConnect'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider. | Use profile ''DE-MK_Network-VoiceConnect'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider. | ||
Please note the following configuration hints: | |||
* Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code' | |||
: {{SIP_TEST_V12_HINT}} | : {{SIP_TEST_V12_HINT}} | ||
== Disclaimer == | == Disclaimer == | ||
{{SIP_TEST_PREFACE}} | {{SIP_TEST_PREFACE}} | ||
Revision as of 17:25, 22 September 2017
Summary
Tests for the VoiceConnect SIP trunk product of the provider MK_Network were completed. Test results have been last updated on September 13th, 2017. Check the history of this article for the date of the first publication of the testreport. <internal>Provider SBC: TELES.iSWITCH</internal>
List of Issues found in media-relay Configuration
- SIP INFO
- The provider does not support conveying DTMF using the SIP-INFO method.
- REDIR 302
- The provider does not support external call redirection using the SIP
302 Redirectresponse - SDP VIDEO
- The provider does not support receiving video media capabilities in the SDP-part of a SIP message.
Here is the list of test-cases that have been performed for this provider: BASIC_CALL, 180_RINGING, CLIR, CLNS, CLNS_ONNET, CONN_NR, CONN_NR_DIFF, EARLY_MEDIA_INBOUND, REVERSE_MEDIA, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, DTMF, G711A, G711A_ONNET, G711U, G711U_ONNET, G722, G722_ONNET, G729, G729_ONNET, HOLD_RETRIEVE, MOBILITY, SIP_INFO, OPUS_NB, OPUS_WB, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, FAX_T38, FAX_AUDIO, FAX_T38_ONNET, FAX_T38ANDAUDIO, SRTP_OUTGOING, SRTP_INCOMING, SRTP_INTERNAL, SDP_VIDEO
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
- 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.
- 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.
- 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.
- 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
- 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)
- 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.
- COLP
- Outbound calls to the PSTN show the correct connected number. However incoming calls from the PSTN do not.
- A caller from the PSTN will receive an incorrect connected number that differs from the dialled number. This might lead to the caller cancelling the call.
- 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.
- Reverse Media Negotiation
- OK
- IP-Fragmentation
- OK
- Large SIP messages
- OK
- Codecs
- supported to/from PSTN: G711A, G711U and G729
- supported onnet (VoIP to VoIP): G711A, G711U and G729
- 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.
- Call Transfer
- OK
- Fax
- Transport of faxes to/from the PSTN via G.711 codec was tested successfully.
- Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. However since the provider requires the MediaRelay and Exclusive-Coder setting, T.38 is not activated on the SIP-interface. The reason for this is the non-working fallback to AudioFax of the Fax-interface, in case that the SIP-interface is configured with above options. This limitation applies only to the Fax-interface. If you are not using it for fax calls, you can enable T.38 by using the "Expert Mode" at the SIP-profile.
- SRTP
- The provider does not support audio encryption using SRTP.
Configuration
Use profile DE-MK_Network-VoiceConnect in Gateway/Interfaces/SIP to configure this SIP provider.
Please note the following configuration hints:
- Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
- A most recent v12r2 firmware or higher is required to use this SIP-profile. For hints regarding upgrade to v12r2, see Howto:Firmware Upgrade V12r1 V12r2
New profiles are added in the course of our V12R2 software Service Releases, see Support:DVL-Roadmap Firmware 12r2. 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.