Howto:CH - Swisscom - Smart Business Connect SIP-Provider (2016)

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Contents

Summary

Tests for the Smart_Business_Connect SIP trunk product of the provider Swisscom were completed. Test results have been last updated on October 19th, 2016. Check the history of this article for the date of the first publication of the testreport.

Remarks

Based on the same firmware architecture the test applies to all Innovaphone gateways.

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 3 (120936)


Note: Based on a report of a partner with installation on 3/2019 there was some issues regarding incoming INVITE from the Cisco SBC to the Innovaphone PBX getting rejected duo the fact of the SIP Register provides a different Contact Header than the one provided on the INVITE. The workaround solution for the issue was to use the option "/support-broken-registrar" and local port 5060 on the SIP interface. This doesn't mean that every installation will have the same issue however we leave this remark here if similar issue happens.

The Flag "/support-broken-registrar" can be set in "Gateway -> SIP -> SIP-Interface -> SIP Interop Tweaks -> Advanced"

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.
MOBILITY 
The provider can not send DTMF signals via SIP-INFO messages.
XFER BLIND 
The provider does not fully support blind call transfer scenarios.
XFER CONS ALERT 
The provider does not fully support consultation call transfer after alert scenarios.
XFER CONS EXT 
The provider does not fully support external consultation call transfer scenarios.
XFER CONS 
The provider does not fully support consultation call transfer after connect scenarios.

Here is the list of test-cases that have been performed for this provider: EARLY_MEDIA_INBOUND, REVERSE_MEDIA, BASIC_CALL, CONN_NR, 180_RINGING, CLIR, CLNS, DTMF, G711A_ONNET, G711U_ONNET, G722_ONNET, G729_ONNET, FAX_AUDIO, FAX_T38, G711A, G711U, G722, G729, HOLD_RETRIEVE, MOBILITY, OPUS_NB, OPUS_WB, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, SRTP_OUTGOING, SRTP_INCOMING, SRTP_INTERNAL, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, SIP_INFO

List of Issues found in media-relay Configuration

This section lists the results that differ from the results for the first configuration.

MOBILITY 
This feature, which does not work 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.
CLIP 
OK
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 not supported by the provider. As a result, you cannot have a Standby gateway/PBX using the same account for failover or load-balancing purposes.
Early-Media 
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
Reverse Media Negotiation 
OK
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.
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. This provider supports call redirection using the SIP 302 Redirect header. Rerouting can be enabled by setting the Reroute supported check-mark in the corresponding Trunk object configuration. Also, the Interworking check-mark must be set in the route used for calls from and to the SIP provider.
However, during our test other interop problems were discovered when the Interworking Flag is enabled. Therefore it is not recommended to use the call redirection via SIP 302 Redirect header. However, on-net (that is, from SIP provider to another customer at the same SIP provider) CLIP no screening (CLNS) is not possible.
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. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.
Codecs 
supported to/from PSTN: G711A, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
Call Transfer 
The provider does not handle internally transferred-after-connect calls.
The provider does not handle internally transferred-after-alert calls.
The provider does not handle internally blind-transferred calls.
The provider does not handle externally transferred calls.
SRTP 
The provider does not support audio encryption using SRTP.
IP-Fragmentation 
OK
Large SIP messages 
OK
Mobility Calls 
OK
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.


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.
CLIP 
OK
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 not supported by the provider. As a result, you cannot have a Standby gateway/PBX using the same account for failover or load-balancing purposes.
Early-Media 
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
Reverse Media Negotiation 
OK
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.
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. This provider supports call redirection using the SIP 302 Redirect header. Rerouting can be enabled by setting the Reroute supported check-mark in the corresponding Trunk object configuration. Also, the Interworking check-mark must be set in the route used for calls from and to the SIP provider.
However, during our test other interop problems were discovered when the Interworking Flag is enabled. Therefore it is not recommended to use the call redirection via SIP 302 Redirect header. However, on-net (that is, from SIP provider to another customer at the same SIP provider) CLIP no screening (CLNS) is not possible.
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. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.
Codecs 
supported to/from PSTN: G711A, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
Call Transfer 
The provider does not handle internally transferred-after-connect calls.
The provider does not handle internally transferred-after-alert calls.
The provider does not handle internally blind-transferred calls.
The provider does not handle externally transferred calls.
SRTP 
The provider does not support audio encryption using SRTP.
IP-Fragmentation 
OK
Large SIP messages 
OK
Mobility Calls 
OK
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.


Configuration

Use profile CH-Swisscom-Smart_Business_Connect 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.

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.

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