Howto:DE - MK Network - VoiceConnect SIP-Provider (2016)

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Summary

Tests for the VoiceConnect SIP trunk product of the provider MK_Network were completed. Test results have been last updated on June 22th, 2021. Check the history of this article for the date of the first publication of the testreport. <internal>Provider SBC: TELES.iSWITCH</internal>

Remarks

The provider doesn't offer the possibility to test the SIP-trunk regularly and automatically (e.g. before firmware releases). As a result, we do not know if it will still work or will work with different firmware than the one we tested with.

Tested Firmware: 13r1 Service Release 28 (133015)

List of Issues found in media-relay Configuration

EARLY MEDIA INBOUND
The provider does not support early-media (i.e. establish RTP-stream before 200 OK/connect) for calls to the PSTN.
MOBILITY
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.
REDIR 302
The provider does not support external call redirection using the SIP 302 Redirect response
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.

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: or Diversion: 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 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 does not support early (that is, before connect) media for outbound calls to the PSTN (hence no inbound early media). This may be an issue in cases where such media is played to the caller (e.g. when calling an unavailable mobile phone).
Reverse Media Negotiation
OK
IP-Fragmentation
OK
Large SIP messages
OK
Codecs
supported to/from PSTN: G711A
supported onnet (VoIP to VoIP): G711A, G711U, G722 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.
Transport of faxes using T.38 worked to onnet destinations. However it failed to PSTN destination and moreover fallback to audiofax failed also.
As a result, T.38 is disabled on the SIP-interface and the use of audio-fax is necessary.
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 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.