Howto:DE - QSC - IPFonie Extended Connect TLS SRTP SIP-Provider (2016)
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 December 21st, 2016. 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.
<internal>Provider SBC: Huawei SoftX3000 V300R010</internal>
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:
orDiversion:
SIP headers for providing the correct calling line id for diverted calls.
- 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.
- The provider does not support T.38 fax calls to the PSTN. You need to use audio-fax therefore.
- Codecs
- supported to/from PSTN: G711A, G711U and G729
- supported onnet (VoIP to VoIP): G711A, 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:
orDiversion:
SIP headers for providing the correct calling line id for diverted calls.
- 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.
- The provider does not support T.38 fax calls to the PSTN. You need to use audio-fax therefore.
- 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.