Howto:IT - VoipVoice - SIP Trunk SIP-Provider (2018)

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Contents

Summary

Tests for the Voip SIP trunk product of the provider VoipVoice were completed. Test results have been last updated on May 4th, 2018. Check the history of this article for the date of the first publication of the testreport.


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: 12r2 Service Release 13


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.

CLIR 
The provider does not fully support suppression of the calling line id (CLIR) using the SIP Privacy: Id header.
CLNS ONNET 
Onnet-Calls (that is, within the provider's network) do not allow foreign calling party numbers (CGPN). In other words, clip no screening is not possible for on-net calls.
CLNS 
Outgoing calls cannot be sent with a foreign calling party number (CLI).
FAX AUDIO 
The provider does not fully support Audiofax (i.e. non-T.38)
FAX T38 
The provider does not fully support T.38 fax
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
REDIR DIVHDR 
The provider does not support maintaining the original caller ID for redirected calls (e.g. CFU) when the original caller is present in a Diversion: header.
REDIR HISTHDR 
The provider does not support maintaining the original caller ID for redirected calls (e.g. CFU) when the original caller is present in a History-Info: header.
SDP VIDEO 
The provider does not support receiving video media capabilities in the SDP-part of a SIP message.
SIP INFO 
The provider does not support conveying DTMF using the SIP-INFO method.

Here is the list of test-cases that have been performed for this provider: CLIR, CLNS_ONNET, CLNS, CONN_NR_DIFF, FAX_AUDIO, FAX_T38, FAX_T38ANDAUDIO, G722_ONNET, G722, OPUS_NB, OPUS_WB, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, SDP_VIDEO, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, RALERT_DISC, SUBSCRIBER_NR, 180_RINGING, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, FAX_T38_ONNET, IP_FRAGMENTATION, LARGE_SIP_MESSAGES

List of Issues found in media-relay Configuration

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

The test results for this configuration are the same, however.

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.
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 can convey DTMF digits using the RTP payload method as per RFC2833. 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 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.
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.
Also, there is no other method available for this provider to make sure externally forwarded calls will show proper calling line identification (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the forwarded call will at least carry the diverting party number (DGPN) as calling party number (CGPN/CLI) when alerting at the forwarded-to target. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
Fax 
Audio-Fax calls (that is, fax calls without T.38) do not work.
Transport of faxes using T.38 worked to onnet destinations. However it failed to PSTN destination. Fallback to audiofax worked.
As a result, T.38 is enabled on the SIP-interface, the use of audio-fax is necessary.
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.
Neither Clip no screening (CLNS) nor call redirection using the SIP Diversion: or History-Info is supported. Calls forwarded to mobility devices will thus not have the original callers calling line id (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the mobility call will at least carry the mobility user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
SRTP 
The provider does not support audio encryption using SRTP.
Dialing of Subscriber Numbers 
OK
Correct signalling of Ringing-state 
OK
Call Transfer 
OK
IP-Fragmentation 
OK
Large SIP messages 
OK


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 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 can convey DTMF digits using the RTP payload method as per RFC2833. 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 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.
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.
Also, there is no other method available for this provider to make sure externally forwarded calls will show proper calling line identification (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the forwarded call will at least carry the diverting party number (DGPN) as calling party number (CGPN/CLI) when alerting at the forwarded-to target. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
Fax 
Audio-Fax calls (that is, fax calls without T.38) do not work.
Transport of faxes using T.38 worked to onnet destinations. However it failed to PSTN destination. Fallback to audiofax worked.
As a result, T.38 is enabled on the SIP-interface, the use of audio-fax is necessary.
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.
Neither Clip no screening (CLNS) nor call redirection using the SIP Diversion: or History-Info is supported. Calls forwarded to mobility devices will thus not have the original callers calling line id (CLI). In this case, we recommend to set the Set Calling = Diverting No check-mark in the PBX Trunk object. This way, the mobility call will at least carry the mobility user's own extension as calling party number (CGPN/CLI) when alerting at the mobile device. Otherwise it would either show no CLI at all or the lowest subscriber number associated with the trunk only.
SRTP 
The provider does not support audio encryption using SRTP.
Dialing of Subscriber Numbers 
OK
Correct signalling of Ringing-state 
OK
Call Transfer 
OK
IP-Fragmentation 
OK
Large SIP messages 
OK


Configuration

Use profile IT-VoipVoice-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

  • STUN required on all endpoints if media-relay is not used
  • 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects
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.

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