Howto:CZ - Radiokomunikace - SIP Trunk SIP-Provider (2017)

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated January 13th, 2017) and may (and probably will) change.

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: 12r1 Service Release 6 (121022)

The use of STUN Server should be used only if the PBX uses Internet to do the SIP Trunk, if there is a direct connection to the Provider Network (via SBC connected to their MPLS) then we can remove the STUN Server setting.

<internal>Provider SBC: Cirpack/v4.58 (gw_sip)</internal>

List of Issues found in media-relay Configuration

CLNS

Outgoing calls cannot be sent with a foreign calling party number (CLI).

CONN NR INCOMING

Incoming calls from the PSTN don't show a correct connected number to the calling party.

FAX T38 ONNET

The provider does not support T.38 fax for onnet calls.

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.

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)

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.

SIP INFO

The provider does not support conveying DTMF using the SIP-INFO method.

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: BASIC_CALL, CLIR, CLNS_ONNET, 180_RINGING, 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, 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 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 cannot handle calls to clients behind NAT. Clients are required to use NAT-traversal methods like STUN. Drawback of this solution is that STUN doesn't work for all NAT routers (i.e. routers doing symmetric NAT). Because of this limitation, it depends on the customer network equipment whether the SIP-trunk is usable or not. However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call.

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.

CLIR

OK

Correct signalling of Ringing-state

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. 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.

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.

Fax

Transport of faxes to/from the PSTN via G.711 codec was tested successfully.

Transport of faxes using T.38 worked to PSTN destinations. However it failed to onnet destination. Fallback to audiofax worked.

As a result, T.38 is enabled on the SIP-interface, the use of audio-fax is necessary.

Codecs

supported to/from PSTN: G711A, G711U and G729

supported onnet (VoIP to VoIP): G711A, G711U and G729

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.

SRTP

The provider does not support audio encryption using SRTP.

Provider supports dialling subscriber numbers

OK

Call Transfer

The provider does not handle internally transferred-after-connect calls.

Configuration

Use profile CZ-Radiokomunikace-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

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.