Howto:NL - KPN - Vast bellen Dienst VoIP Connect SIP-Provider (3rdParty 2017)

Summary

Tests for the Vast_bellen_Dienst_VoIP_Connect SIP trunk product of the provider KPN were completed. Test results have been last updated on August 25th, 2017. Check the history of this article for the date of the first publication of the testreport.

However the tested SIP trunk is not available for nightly tests yet! 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 3 (125210)

Remarks

• The testresults are not from an automatic test. The test is done manual on a KPN location in Houten(NL) in a dedicated test envirement.

The test desciption is mainly based on the MEDIA RELAY configuration but there also some tests performed with direct RTP. If you want to use direct RTP configuration instead of MEDIA RELAY configuration, you need to configure all endpoints with Codec G711A exclusive or only with the by KPN supported codecs (G711 and G722)

Certification Status and test results

This product is listed due to a customer testimonial. No tests have been conducted by innovaphone.

Testing of this product has been finalized 23 August , 2017.

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 in the codec offer from 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. The provider has a 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 certain duration (i.e default SIP-registration interval) incoming and outgoing calls might be rejected/fail.|timeout=1 minutes}}

The KPN SIP trunk is a SIP trunk without registration. The availability is checked with the SIP Options and if configured as TCP also on OSI-layer 3.

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)

CLIP no screening (CLNS) is not possible, that is, outgoing calls can only have the own subscriber number (or extension), no foreign numbers as calling line id. This affects for example externally forwarded and mobility calls. However, on-net (that is, from SIP provider to another customer at the same SIP provider) CLIP no screening (CLNS) is possible. As this is a non-German provider, the issue with off-net CLNS could be related to the provider or related to the international PSTN peering. Please consult the SIP provider if CLNS will work for you.

According to Dutch regulation for outgoing calls, it is not allowed to present a telephone number other than a number that belongs tot he customer.

With external diversion or forwarding it is allowed to present a foreign number (A-number) with the use of Diversion Header.

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

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 and G722

supported onnet (VoIP to VoIP): G711A, G711U and G722

IP-Fragmentation

OK

Large SIP messages

OK

Reverse Media Negotiation

Reverse-media negotiation is not supported. Therefore, a media-relay with exclusive coder configuration will be activated on the SIP-interface.

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.

SRTP

The provider does not support audio encryption using SRTP.

Provider supports dialling subscriber numbers

OK

Call Transfer

OK

Configuration

Use profile NL-KPN-Vast_bellen_Dienst_VoIP_Connect 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.