Howto:DE - PYUR Business - SIP Trunk SIP-Provider (2021): Difference between revisions
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; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template: | ; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ||
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}} | ; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}} |
Revision as of 15:08, 3 May 2021
Summary
PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated May 3rd, 2021) and may (and probably will) change. <internal>Provider SBC: </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
- The provider does not fully support T.38 fax
- MOBILITY
- The provider can not send DTMF signals via SIP-INFO messages.
- 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_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.
- 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 UDP and TCP as transport protocol. The tests were completed using TCP, since UDP is an unreliable protocol and 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.
- 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. The provider has a 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 a certain duration (i.e default SIP-registration interval) incoming and outgoing calls might be rejected/fail.|timeout=2 minutes}}
- 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. 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.
- 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 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.
- Codecs
- supported to/from PSTN: G711A
- supported onnet (VoIP to VoIP): G711A, G711U and G722
- 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 does not support audio encryption using SRTP.
- Dialing of Subscriber Numbers
- OK
- Call Transfer
- OK
Configuration with media-relay
- Registration
- The provider supports UDP and TCP as transport protocol. The tests were completed using TCP, since UDP is an unreliable protocol and 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.
- 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. The provider has a 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 a certain duration (i.e default SIP-registration interval) incoming and outgoing calls might be rejected/fail.|timeout=2 minutes}}
- 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. 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.
- 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 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.
- Codecs
- supported to/from PSTN: G711A
- supported onnet (VoIP to VoIP): G711A, G711U and G722
- 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 does not support audio encryption using SRTP.
- Dialing of Subscriber Numbers
- OK
- Call Transfer
- OK
Configuration
Use profile DE-PYUR_Business-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.
- 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.