Howto:DE - Deutsche Telekom - DeutschlandLAN SIP Trunk SIP-Provider (2016): Difference between revisions
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== Summary == | == Summary == | ||
{{Template:SIP_TEST_STATUS_complete|update= | {{Template:SIP_TEST_STATUS_complete|update=June 8th, 2018|url=https://geschaeftskunden.telekom.de/startseite/festnetz-internet/tarife/333506/deutschlandlan-sip-trunk.html|productname=DeutschlandLAN_SIP_Trunk|providername=Deutsche_Telekom}} | ||
<internal>Provider SBC: unknown</internal> | <internal>Provider SBC: unknown</internal> | ||
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; FAX T38 : {{SIP_TEST_FACT_FAX T38}} | ; FAX T38 : {{SIP_TEST_FACT_FAX T38}} | ||
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}} | ; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}} | ||
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}} | ; SIP INFO : {{SIP_TEST_FACT_SIP INFO}} | ||
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; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}} | ; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}} | ||
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template: | ; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_optional}} | ||
; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_no}}{{Template: | ; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_no}} {{Template:SIP_Profile_Test_COLP_diff_no}} | ||
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}} | ; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}} | ||
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} | ||
: {{Template: | : {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}} | ||
; Codecs : supported to/from PSTN: G711A and G729 | ; Codecs : supported to/from PSTN: G711A and G729 | ||
| Line 48: | Line 46: | ||
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}} | ; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}} | ||
; Reverse Media Negotiation : {{Template: | ; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}} | ||
; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} | ; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} | ||
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Please note the following configuration hints: | Please note the following configuration hints: | ||
* Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code' | * <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki> | ||
: {{SIP_TEST_V12_HINT}} | : {{SIP_TEST_V12_HINT}} | ||
Revision as of 10:43, 8 June 2018
Summary
Tests for the DeutschlandLAN_SIP_Trunk SIP trunk product of the provider Deutsche_Telekom were completed. Test results have been last updated on June 8th, 2018. Check the history of this article for the date of the first publication of the testreport. <internal>Provider SBC: unknown</internal>
List of Issues found in media-relay Configuration
- FAX T38
- The provider does not fully support T.38 fax
- FAX T38ANDAUDIO
- The provider does not support fallback to audio-fax if T.38 fails.
- 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, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38, FAX_T38ANDAUDIO, FAX_T38_ONNET, 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, SDP_ICE, SDP_RTCP_MUX, SDP_VIDEO
Test Results
This SIP provider requires a media-relay configuration. That is, all media traffic between the SIP provider and all endpoints must flow through the SBC. For this reason, a configuration without media-relay has not been tested and hence no test results for this configuration are listed.
Configuration with media-relay
- Registration
- The provider supports only TCP as transport protocol. In general TCP is preferred to UDP, 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. 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.
- Session Timer
- The tests regarding the SIP-session timer were not successful. This will result in unwanted call termination on calls exceeding a certain time (default 30 minutes). Because of this, further tests were aborted.
- 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 240 seconds 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. 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, this configuration is not strictly required, as the provider supports clip no screening so that redirected calls (i.e. call forwards to external numbers) will show a proper calling line id (CLI) at the receiving party anyway. However, it may be useful anyhow to get rid of externally forwarded calls on the SBC entirely.
- 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.
- 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 worked to onnet destinations. However it failed to PSTN destination and moreover fallback to audiofax 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 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 does not support audio encryption using SRTP.
- Call Transfer
- OK
Configuration
Use profile DE-Deutsche_Telekom-DeutschlandLAN_SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.
Please note the following configuration hints:
- Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
- 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.