Howto:DE - Deutsche Telefon - SIP Tk Anlagenanschluss SIP-Provider (2016): Difference between revisions

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== Summary ==
== Summary ==  
The tests for the [http://www.deutsche-telefon.de/sip-trunking.html SIP Tk-Anlagenanschluss] of the provider [http://www.deutsche-telefon.de/ Deutsche Telefon] were completed.
The tests for the ''[http://www.deutsche-telefon.de/sip-trunking.html SIP_Tk_Anlagenanschluss]'' SIP trunk product of the provider ''Deutsche_Telefon'' were completed.


Issues found were:
Test results for this product have been last updated August 18th, 2016.
; SRTP
; NAT Traversal
; Redundancy
; Correct signalling of Ringing-state


For more details about this issues, see the respective test-results sections.


== Current test state ==


{{Template:Compat Status "tested"(sip provider)}}
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}


Testing of this product has been finalized June 10th, 2016.
<small>{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]]</small>
<!-- [[Category:RecProd|{{PAGENAME}}]] -->
<!-- [[Category:3rdParty SIP Provider|{{PAGENAME}}]] -->


== Tests with MediaRelay ==
; Registration : The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted. Moreover it requires all involved network elements to support IP-fragmentation.


; CLIP : OK
== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}


; CLIR : OK
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}


; Clip No Screening(CLNS) : OK
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}
: Additionally, redirection using "302 Moved Temporary" is possible. In case of a call forward, the incoming call is redirected to the PSTN. From point of view of the PBX the call doesn't exist after the redirection and is handled by the SIP-Provider. This is equivalent to "Partial Rerouting/Call Deflection" in ISDN. When a redirection is done, the redirection target will receive the original calling number, so for forwarded calls this will have the same effect as CLNS. However this redirection is not usable for Mobility calls or plain CLNS-calls. A plain CLNS-call is an outbound call with a "wrong" CGPN number (i.e. one that does not belong to the trunk). This happens e.g. if a call-center wants to send a service number (such 0800xxx) as CLI.


; Codecs : The provider support the following codecs: G711A
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}


; Fax : Transport of faxes to/from the PSTN via G.711(A/U) codec was tested successfully. Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}


; SRTP : The provider does not support audio encryption using SRTP.
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}


; DTMF (RFC2833) : OK
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}


; 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 expects that all RTP-packets are passed through the PBX. 
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_optional}}


; Reverse Media Negotiation : OK
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_yes}}


; Mobility Call : Transmitting DTMF-tones as SIP-INFO messages was not possible, however we can do Mobility-calls since we use MediaRelay on the SIP-Interface.
; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A


; 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 a duration of up to 2 minutes (i.e registration interval) incoming and outgoing calls might be rejected/fail.
; Mobility Call :  {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}


; IP-Fragmentation : OK
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}


; Large SIP messages : OK
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}


; Correct signalling of Ringing-state : Ringing is not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing. <br>Additionally external callers forwarded/transferred back to the PSTN, will get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call.
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}


; Early-Media : The provider supports early-media for outbound calls to the PSTN.
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}


; 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.
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}
 
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}


; Call Transfer : OK


== Tests without MediaRelay==
The tests without MediaRelay were aborted, since it is required by the provider. The reason for it, are audio problems when two external RTP-endpoints are connected(e.g. external transfer, mobility call).


==Configuration==
==Configuration==
* Use profile ''DE-Deutsche_Telefon-SIP_Tk_Anlagenanschluss'' in the Gateway/Interfaces/SIP menu.
Use profile ''DE-Deutsche_Telefon-SIP_Tk_Anlagenanschluss'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
 
==Contact==


[http://www.deutsche-telefon.de/kontakt.html contact form of provider]
== Disclaimer ==
{{SIP_TEST_PREFACE}}


[[Category:Compat|{{PAGENAME}}]]
[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Revision as of 10:44, 18 August 2016

Summary

The tests for the SIP_Tk_Anlagenanschluss SIP trunk product of the provider Deutsche_Telefon were completed.

Test results for this product have been last updated August 18th, 2016.


List of Issues found in media-relay Configuration

180 RINGING
The provider does not send a 180 Ringing response when the called party alerts.
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, DTMF, FAX_AUDIO, FAX_T38, G711A, G711U, G722, G729, HOLD_RETRIEVE, MOBILITY, SIP_INFO, OPUS_NB, OPUS_WB, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, SRTP_OUTGOING, SRTP_INCOMING, SRTP_INTERNAL, CONN_NR_DIFF, CONN_NR, EARLY_MEDIA_INBOUND, G711A_ONNET, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, RALERT_DISC, REVERSE_MEDIA


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

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.
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.
Registration
The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted(TLS). Moreover it requires all involved network elements to support IP-fragmentation.
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.
DTMF (RFC2833)
The provider can convey DTMF digits using the RTP payload method as per RFC2833.
Correct signalling of Ringing-state
Ringing is not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing.
Additionally external callers forwarded/transferred back to the PSTN, may get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call. As a result, Carrier w/o Alerting is required in all PBX 'Mobility' objects.
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. 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.
Fax
Transport of faxes to/from the PSTN via G.711 codec was tested successfully.
Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.
Codecs
supported to/from PSTN: G711A
supported onnet (VoIP to VoIP): G711A
Mobility Call
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.
Call Transfer
OK
SRTP
The provider does not support audio encryption using SRTP.
Early-Media
The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
IP-Fragmentation
OK
Large SIP messages
OK
Reverse Media Negotiation
OK


Configuration

Use profile DE-Deutsche_Telefon-SIP_Tk_Anlagenanschluss in Gateway/Interfaces/SIP to configure this SIP provider.

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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