Howto:ES - Vozelia - Telefonia IP SIP-Provider (2017)

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Contents

Summary

Tests for the Telefonia_IP SIP trunk product of the provider Vozelia were completed. Test results have been last updated on August 10th, 2017. Check the history of this article for the date of the first publication of the testreport.

Remarks

  • SRTP: The provider supports audio encryption using SRTP for incoming, outgoing and on-net calls. To use it you have to contact Vozelia to activate it. In the template it is not activated.
  • TLS For using TLS you have to contact Vozelia and activate it. Till now we didn't test it.
  • Fax For Fax Vozelia offers G711 on a seperate trunk\product.
  • DTMF It can be that on international calls sometimes DTMF tones don't work correct. You can contact Vozelia and it will be fixed.


List of Issues found in media-relay Configuration

180 RINGING 
The provider does not send a 180 Ringing response when the called party alerts.
FAX AUDIO 
The provider does not fully support Audiofax (i.e. non-T.38)
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.
HOLD RETRIEVE 
The provider does not support Hold/Retrieve of a call.
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.
RALERT DISC 
Call disconnected by far end during alert does not disconnect locally
REDIR 302 
The provider does not support external call redirection using the SIP 302 Redirect response

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, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, SUBSCRIBER_NR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS


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 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. However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call.
DTMF (RFC2833) 
The provider does not support conveying telephony events (a.k.a. DTMF) in-band as RTP payload. However, it supports conveying telephony events (DTMF) using the SIP-INFO method. innovaphone endpoints will work with this method too. 3rd party endpoints may have issues though.
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 
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 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 supports early-media for outbound calls (hence inbound early media) to the PSTN.
Fax 
Audio-Fax calls (that is, fax calls without T.38) do not work. However, all fax endpoints must be configured with exclusive codec "G711A".
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 and G729
IP-Fragmentation 
OK
Large SIP messages 
OK
Reverse Media Negotiation 
OK
Mobility Calls 
OK
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.
SRTP 
The provider does not support audio encryption using SRTP.
Provider supports dialling subscriber numbers 
OK
Call Transfer 
OK


Configuration

Use profile ES-Vozelia-Telefonia_IP in Gateway/Interfaces/SIP to configure this SIP provider.

  • Profiles are added in the course of our V12R2 software Service Releases, see Support:DVL-Roadmap Firmware 12r2. Note: You can search in the article page using the key words "SIP-Provider Profile"

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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