General settings WLVoIP

Our telephony platform wlvoip.net uses the SIP-standard of RFC3261.  UDP and TCP-transport is supported. Over IPv4 and IPv6. We do not support the STCP (or TLS) protocol. The following are the general SIP settings that should be used with our WLVoip platform:


Host for proxy and registration: 

sip.wlvoip.net
Generic

on port 5060 (UDP or TCP)

Host for STUN (only for IPv4 used with NAT):

sip.wlvoip.net
HTML

Domain (oftenly called realm):

sip.wlvoip.net
HTML


The username and password can be found on the delivery info note (in dutch Opleverbrief). The username, call forward and caller ID can be viewed and managed on our self service portal: https://my.signetbreedband.nl


Algorithms for speech (codec)

Voice channels can be used with the following algorithms:

  • ITU G.711 A-law (PCM)

  • ITU G.711 μ-law (PCM)

  • ITU G.729A

  • ETSI GSM

Tone dialing (DTMF)

Tone dialing is supported as described in RFC2833, this is also known as SIP-INFO.

Number display (CLIP/CLIR)

Be aware that the following settings can only be used if the Display number setting, is set to "manually" in our self service portal https://my.signetbreedband.nl.

After this it is possible to set a caller-id in the CLIP-setting. This can be done in the following formats. (see incoming number format section).
The P-Asserted-Identity of P-Preferred-Identity SIP header (RFC3325) should be used. This can be combined if necessary with a Privacy SIP header (RFC3323). As a alternative the From SIP header can be used.
Next, a example of the P-Asserted-Identity header, with or without the Privacy header:

P-Asserted-Identity without privacy header:

"0201234567" <sip:0201234567@sip.wlvoip.net>

P-Asserted-Identity with privacy header:

"0201234567" <sip:0201234567@sip.wlvoip.net> Privacy: none

Below examples of a CLIR with p-preferred-identity header, with and without privacy header, and of the "From" header, also with a anonymous example.

Example of the P-Preferred-Identity without privacy header: 

"Anonymous" <sip:anonymous@anonymous.invalid>


Example of the P-Preferred-Identity with privacy header:

"0201234567" <sip:0201234567@sip.wlvoip.net> Privacy: id

Example of the From header:

"0201234567" <sip:username@sip.wlvoip.net>


Example of a anonymous From header:

"anonymous" <sip:username@sip.wlvoip.net>


Important note! Please be aware that if our wlvoip platform is not able to recognize the specific format, calls will be converted as "anonymous" and will not display a caller-id at all.


Outbound number format

The following outbound number formats are valid:

  • Dutch national phone number format (10 digits), for example:

    0201234567

  • International format with two zero prefix, for example:

    0031201234567

  • international format with + prefix, for example:
    +31201234567


Inbound number format

By default, inbound calls will be delivered to the registered SIP UA (phone/PBX). In case you would like to route (DDI/DID) calls, these can be done in the following formats:

  • Dutch national phone number format without zero prefix, for example:
    201234567

  • Dutch national phone number format with zero prefix, for example:

    0201234567

  • International phone number format dubble zero prefix, for example: 

    0031201234567
  • International phone number format with + prefix, for example:
    +31201234567

Important note! Please be aware that if our wlvoip platform is not able to recognize the specific format, calls will be converted as "anonymous" and will not display a caller-id at all.


Call forwarding

If you need to set a call forwarding for your phone or PBX. The system supports a "302 Moved temporarily" setting.
As an alternative, for example, if the PBX is not available, this forwarding state can be set in our self service portal. https://my.signetbreedband.nl

Network traffic filtering

If you need to filter network traffic and setup access lists in your firewall, please use the following IP subnet:

  • IPv4:
    31.223.168.224/27
  • IPv6: 
    2001:4cb8:34a:10::/64

The following ports should be allowed:

  • SIP signaling: UDP or TCP port 5060
  • RTP traffic: UDP ports 10000 to 20000

Not supported note

The following standards and products will not be supported:

  • SIP REFER (RFC3515)
  • SIP re-INVITE (part of the RFC3261) 
  • Message Waiting Indication (RFC3842) Video (part of the RFC3840)
  • Microsoft Lync (all versions)
  • ITU T.38 (for fax over IP)