Howto:DE - Plusnet - IPFonie Extended Connect TLS SRTP SIP-Provider (2016): Difference between revisions
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== Summary == | == Summary == | ||
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2020|url=https://www.plusnet.de/de/corporate-solutions/telefonie/sip-anschluesse/ipfonie-extended-connect/|productname= | {{Template:SIP_TEST_STATUS_complete|update=April 22th, 2020|url=https://www.plusnet.de/de/corporate-solutions/telefonie/sip-anschluesse/ipfonie-extended-connect/|productname=IPfonie_Extended_Connect_TLS_SRTP|providername=Plusnet}} | ||
=== Remarks === | === Remarks === | ||
* Registration: The provider supports also unencryted communication with SIP-TCP/RTP. Tests were done only with encryption(SIP-TLS/SRTP). | * Registration: The provider supports also unencryted communication with SIP-TCP/RTP. Tests were done only with encryption(SIP-TLS/SRTP). | ||
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: {{SIP_TEST_FACT__unreliable}} | : {{SIP_TEST_FACT__unreliable}} | ||
; XFER CONS : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}} | ; XFER CONS : {{SIP_TEST_FACT_WORKSINALTERNATE_UNSTABLE_IN_PRIMARY}} | ||
== Test Results == | == Test Results == |
Revision as of 12:10, 27 April 2020
Summary
Tests for the IPfonie_Extended_Connect_TLS_SRTP SIP trunk product of the provider Plusnet were completed. Test results have been last updated on April 22th, 2020. Check the history of this article for the date of the first publication of the testreport.
Remarks
- Registration: The provider supports also unencryted communication with SIP-TCP/RTP. Tests were done only with encryption(SIP-TLS/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.
- EARLY MEDIA INBOUND
- The provider does not support early-media (i.e. establish RTP-stream before 200 OK/connect) for calls to the PSTN.
- FAX T38 ONNET
- The provider does not support T.38 fax for onnet calls.
*Like written in the remarks this test fails duo the encryption.
- FAX T38
- The provider does not fully support T.38 fax
*Like written in the remarks this test fails duo the encryption.
- FAX T38ANDAUDIO
- The provider does not support fallback to audio-fax if T.38 fails.
*Like written in the remarks this test fails duo the encryption.
- REDIR 302
- The provider does not support external call redirection using the SIP
302 Redirect
response - 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: 180_RINGING, BASIC_CALL, CLIR, CLNS_ONNET, 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, SDP_ICE, SDP_RTCP_MUX, SDP_VIDEO, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, SUBSCRIBER_NR, 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.
- G711A ONNET
- The provider does not support the G711A codec for on-net 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.
- XFER CONS EXT
- The provider does not fully support external consultation call transfer 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.
- XFER CONS
- This feature, which is unstable 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.
- 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.
- COLP
- Outbound and inbound calls to/from the PSTN show the correct connected number.
- For outbound calls to the PSTN, an update of the connected number is not signalled to the caller.
- Early-Media
- The provider does not support early (that is, before connect) media for outbound calls to the PSTN (hence no inbound early media). This may be an issue in cases where such media is played to the caller (e.g. when calling an unavailable mobile phone).
- 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. Moreover fallback to audio-fax failed also.
- As a result, T.38 is disabled on the SIP-interface and the use of audio-fax is necessary.
- Codecs
- supported to/from PSTN: G711A
- supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
- IP-Fragmentation
- OK
- Large SIP messages
- OK
- Reverse Media Negotiation
- OK
- Mobility Calls
- OK
- SRTP
- The provider supports audio encryption using SRTP for incoming, outgoing and on-net calls.
- Dialing of Subscriber Numbers
- The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
- Call Transfer
- The provider does not handle internally transferred-after-connect calls.
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.
- 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.
- COLP
- Outbound and inbound calls to/from the PSTN show the correct connected number.
- For outbound calls to the PSTN, an update of the connected number is not signalled to the caller.
- Early-Media
- The provider does not support early (that is, before connect) media for outbound calls to the PSTN (hence no inbound early media). This may be an issue in cases where such media is played to the caller (e.g. when calling an unavailable mobile phone).
- 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. Moreover fallback to audio-fax failed also.
- As a result, T.38 is disabled on the SIP-interface and the use of audio-fax is necessary.
- Codecs
- supported to/from PSTN: G711A
- supported onnet (VoIP to VoIP): G711U
- IP-Fragmentation
- OK
- Large SIP messages
- OK
- Reverse Media Negotiation
- OK
- Mobility Calls
- OK
- SRTP
- The provider supports audio encryption using SRTP for incoming, outgoing and on-net calls.
- Dialing of Subscriber Numbers
- The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
- Call Transfer
- The provider does not handle externally transferred calls.
Configuration
Use profile DE-Plusnet-IPFonie_Extended_Connect_TLS_SRTP in Gateway/Interfaces/SIP to configure this SIP provider.
Please note the following configuration hints:
- If you intend to use SIPS (SIP/TLS) registration, you need to add the ' GeoTrust TLS RSA CA G1' certificate to the trust list of your SBC
- Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
- A most recent v13r3 firmware is required to use this SIP-profile. For hints regarding upgrade to v13r3, see Howto:V13_Firmware_Upgrade_V13r2_V13r3
New profiles are added in the course of our V13R3 software Service Releases, see Reference13r3:Release Notes Firmware. 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.