Howto:AT - Russmedia IT - highspeed Telefon SIP-Provider (2016): Difference between revisions

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== Summary ==
== Summary ==  
The tests for the ''highspeed Telefon'' of the provider [http://highspeed.vol.at/russmedia-it/ Russmedia IT GmbH] were completed successfully.
The tests for the ''highspeed_Telefon'' SIP trunk product of the provider [http://highspeed.vol.at/kontakt/ ''Russmedia_IT''] were completed.


Issues found were:
Testing of this product has been finalized August 5th, 2016.
; SRTP
; NAT Traversal
; Mobility Call
; Redundancy


=== Remarks ===
das ist das haus vom nikolaus


For more details about this issues, see the respective test-results sections.
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
== Current test state ==
{{SIP_TEST_ISSUES_MR_INTRO}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}


<!--{{Template:Compat Status "planned"}} -->
<!-- {{Template:Compat Status "in progress"}} -->
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"engineered_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"rec._prod."|certificate=HFO_NGN_Connect_HFO_SIP_Provider_-_product-cert.pdf}} -->
{{Template:Compat Status "tested"(sip provider)}}
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


Testing of this product has been finalized May 10th, 2016.
== Test Results ==
[[Category:RecProd|{{PAGENAME}}]]
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}


== Tests with MediaRelay ==
; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}
; 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
; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}


; CLIR : OK
; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}


; Clip No Screening(CLNS) : OK
; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}


; Codecs : The provider support the following codecs: G711A The following codecs are not supported: G711U, G729, G722, Opus.
; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}


; 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.
; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}


; SRTP : The provider does not support audio encryption using SRTP.
; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}


; DTMF (RFC2833) : OK
; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_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.
; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_yes}}
: {{Template:SIP_Profile_Test_FALLBACKFAX_PSTN_yes}}


; Reverse Media Negotiation : OK
; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A


; Mobility Call : Transmitting DTMF-tones as SIP-INFO messages was not possible, which is required in-order to use Mobility-calls without MediaRelay on the SIP-Interface.
; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}


; 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.
; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}


; IP-Fragmentation : OK
; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}


; Large SIP messages : OK
; Mobility Call : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}


; Correct signalling of Ringing-state : OK
; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}


; Early-Media : The provider supports early-media for outbound calls to the PSTN.
; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}


; 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.


; 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 (e.g. ''AT-Russmedia_IT-highspeed_Telefon'') in the Gateway/Interfaces/SIP menu.
Use profile ''AT-Russmedia_IT-highspeed_Telefon'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.


==Contact==
==Contact==


[http://highspeed.vol.at/kontakt/ provider contact form]
[http://highspeed.vol.at/kontakt/ Kontakt > highspeed Internet]
 
== Disclaimer ==
{{SIP_TEST_PREFACE}}
 
[[Category:Compat|{{PAGENAME}}]]
[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Revision as of 12:19, 5 August 2016

Summary

The tests for the highspeed_Telefon SIP trunk product of the provider Russmedia_IT were completed.

Testing of this product has been finalized August 5th, 2016.

Remarks

das ist das haus vom nikolaus

List of Issues found in media-relay Configuration

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.


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 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.
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 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: or Diversion: SIP headers for providing the correct calling line id for diverted calls.
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.
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.
Fallback from T.38 to G.711 was also tested successfully.
Codecs
supported to/from PSTN: G711A
supported onnet (VoIP to VoIP): G711A
IP-Fragmentation
OK
Large SIP messages
OK
Reverse Media Negotiation
OK
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.
SRTP
The provider does not support audio encryption using SRTP.
Call Transfer
OK


Configuration

Use profile AT-Russmedia_IT-highspeed_Telefon in Gateway/Interfaces/SIP to configure this SIP provider.

Contact

Kontakt > highspeed Internet

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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