Howto:IT - Made in Lab - VoIP In Lab SIP-Provider (2016): Difference between revisions

From innovaphone wiki
Jump to navigation Jump to search
No edit summary
 
Line 3: Line 3:
=== Remarks ===
=== Remarks ===
* Call Signaling: During our test we encountered the issue, that calls from the PSTN(Germany/Vodafone) sporadically take a very long time(>20 seconds) to be signaled at the PBX. The reason for this is unknown, national calls or calls from other countries/PSTN carriers might not have this delay problem. Because of it, some transfer tests(XFER CONS ALERT & XFER CONS ) failed.
* Call Signaling: During our test we encountered the issue, that calls from the PSTN(Germany/Vodafone) sporadically take a very long time(>20 seconds) to be signaled at the PBX. The reason for this is unknown, national calls or calls from other countries/PSTN carriers might not have this delay problem. Because of it, some transfer tests(XFER CONS ALERT & XFER CONS ) failed.
{{Template:SIP_TEST_NO_NIGHTLY_TESTS|fw-version= 12r2 Service Release 21}}


<internal>Provider SBC: Sippy</internal>
<internal>Provider SBC: Sippy</internal>


=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
=== {{SIP_TEST_ISSUES_MR_TITLE}} ===

Latest revision as of 11:49, 23 July 2021

Summary

Tests for the VoIP_In_Lab SIP trunk product of the provider Made_in_Lab were completed. Test results have been last updated on January 13th, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

  • Call Signaling: During our test we encountered the issue, that calls from the PSTN(Germany/Vodafone) sporadically take a very long time(>20 seconds) to be signaled at the PBX. The reason for this is unknown, national calls or calls from other countries/PSTN carriers might not have this delay problem. Because of it, some transfer tests(XFER CONS ALERT & XFER CONS ) failed.


The provider doesn't offer the possibility to test the SIP-trunk regularly and automatically (e.g. before firmware releases). As a result, we do not know if it will still work or will work with different firmware than the one we tested with.

Tested Firmware: 12r2 Service Release 21

<internal>Provider SBC: Sippy</internal>

List of Issues found in media-relay Configuration

180 RINGING
The provider does not send a 180 Ringing response when the called party alerts.
REDIR 302
The provider does not support external call redirection using the SIP 302 Redirect response
XFER CONS ALERT
The provider does not fully support consultation call transfer after alert 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
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.
REVERSE MEDIA
The provider does not support reverse media negotiation (a.k.a. late SDP)
SIP INFO
The provider does not support conveying DTMF using the SIP-INFO method.
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: BASIC_CALL, 180_RINGING, CLIR, CLNS, CLNS_ONNET, CONN_NR, CONN_NR_DIFF, DTMF, EARLY_MEDIA_INBOUND, G711A, G711A_ONNET, G711U, G711U_ONNET, G722, G722_ONNET, G729, G729_ONNET, HOLD_RETRIEVE, MOBILITY, OPUS_NB, OPUS_WB, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS, XFER_BLIND, FAX_T38, FAX_T38_ONNET, FAX_T38ANDAUDIO, FAX_AUDIO, SRTP_OUTGOING, SRTP_INCOMING, SRTP_INTERNAL, REVERSE_MEDIA, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, SIP_INFO


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.
An outgoing call that is disconnected by the far end during alert is not disconnected locally. Typically, the provider may play a message to the effect that the call has been rejected or the call can not be completed so that the calling user will hang up. This may be OK, however, it may create issues with automated calls (e.g. fax or modem) which do not listen to the announcement and keep waiting for the far end to accept the call.
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.
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.
Codecs
supported to/from PSTN: G711A, G711U and G729
supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729
Call Transfer
The provider does not handle internally transferred-after-connect calls.
The provider does not handle internally transferred-after-alert calls.
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.
SRTP
The provider does not support audio encryption using SRTP.
Reverse Media Negotiation
Reverse-media negotiation is not supported. Therefore, a media-relay with exclusive coder configuration will be activated on the SIP-interface.
IP-Fragmentation
OK
Large SIP messages
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.
As the provider does not provide proper alert signalling, you will need to configure the Carrier w/o Alerting check-mark in the PBX Mobility object.


Configuration

Use profile IT-Made_in_Lab-VoIP_In_Lab in Gateway/Interfaces/SIP to configure this SIP provider.

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.