Howto:DE - QSC - IPFonie Extended Connect TLS SRTP SIP-Provider (2016)

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Contents

Summary

Tests for the IPFonie_Extended_Connect_TLS_SRTP SIP trunk product of the provider QSC were completed. Test results have been last updated on January 13th, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

  • Registration: The provider supports also unecryted communication with SIP-TCP/RTP. Tests were done only with encryption(SIPS/SRTP).
  • FAX T38: Encrypted Trunks do not support T.38, because fallback to unencrypted media is disabled for security reasons. T.38 protocol itself does not have an encrypted variant currently specified. Use G.711 pass-through for FAX support.
  • Redundancy: The provider has an own failover detection using OPTIONS-packets. If more the one SIP-Interface are registered at the same account and one interface stops answering the OPTIONS-packet of the provider, calls are not sent to this interface unless OPTIONS-packets are again answered.
    • The OPTIONS-packets intervall is 60 seconds, which is also the max. downtime of not forwarded calls to a working SIP-interface.



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.

SUBSCRIBER NR 
The provider does not support dialling numbers in subscriber number format.
FAX T38 
The provider does not fully support T.38 fax
FAX T38 ONNET 
The provider does not support T.38 fax for onnet calls.
MOBILITY 
The provider can not send DTMF signals via SIP-INFO messages.
REDIR 302 
The provider does not support external call redirection using the SIP 302 Redirect response
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: 180_RINGING, BASIC_CALL, CLIR, CLNS_ONNET, CLNS, CONN_NR_DIFF, CONN_NR, SUBSCRIBER_NR, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38, FAX_T38_ONNET, 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, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS

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 TLS as transport protocol. In general TLS is preferred to TCP or UDP, since it offers encryption of the transmitted SIP-packets.
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 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
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 not signalled to the caller.
Provider supports dialling subscriber numbers 
The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
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 failed to PSTN and onnet destinations. Fallback to audio-fax 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, G722 and G729
IP-Fragmentation 
OK
Large SIP messages 
OK
Reverse Media Negotiation 
OK
Mobility Calls 
Transmitting DTMF-tones as SIP-INFO messages is not supported. In a no-media-relay configuration, DTMF signalling can thus not be conveyed to the PBX. Mobility calls will not work.
SRTP 
The provider supports audio encryption using SRTP for incoming, outgoing and on-net calls.
Call Transfer 
OK


Configuration with media-relay

Registration 
The provider supports only TLS as transport protocol. In general TLS is preferred to TCP or UDP, since it offers encryption of the transmitted SIP-packets.
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 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
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 not signalled to the caller.
Provider supports dialling subscriber numbers 
The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
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 failed to PSTN and onnet destinations. Fallback to audio-fax 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, G722 and G729
IP-Fragmentation 
OK
Large SIP messages 
OK
Reverse Media Negotiation 
OK
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 supports audio encryption using SRTP for incoming, outgoing and on-net calls.
Call Transfer 
OK


Configuration

Use profile DE-QSC-IPFonie_Extended_Connect_TLS_SRTP 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.

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