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WAN VoIP Services

WanCom is the wan internal phonesystem, which allows free calls between members connected to the wan.

The following prefixes have been assigned:

  • 032 - Galway
  • 033 - Clare
  • 034 - Limerick
  • 035 - Wexford
  • 036 - Waterford

These prefixes are not in use in the public phone system in Ireland, so we should not have any problem with these.

To request an account PM Marlow on the forums with the username and the phonenumber that you want. The phonenumber is the prefix for your area and 6 digits.

We can also publish regular phonenumbers on the Wan (091-XXXXXX, 061-XXXXXX, 086-XXXXXXX, 087-XXXXXXX), so if you want that number to be "Wan"-reachable, it can be included in the PM. We will confirm that you own the number and then add it to the system.

The conference center can be found at 1888 (Conference rooms 100-109 are freely accessible).

For help with the voip setup, try to contact me (Marlow) on irc.thewan.net, port 6667, channel #wan, PM me on the forums or check for me in the jabber conferencing (talk.thewan.net) room "wan".


To understand the difference between IAX2 and SIP please read this. Also important for sound quality is what codec you choose. Here is a list of codecs, how much bandwidth they use and what quality you in general get of them.


IAX2 or the Inter Asterisk eXchange protocol Version 2 was designed to cope with firewalls and the like, it uses one port only and transfers both signalling and mediastream in the same connection.


SIP or the Session Initiation protocol uses multiple ports, one for signalling and one or more for the mediastream (voice). This makes it sometimes difficult to get it working with firewalls.


Codecs are used for compression of the mediastream, our conversation basically. The more compression a codec uses, the more it might affect the quality of the call, on the other side, the more bandwidth a codec uses, the more choppy a conversation can be, given the bandwidth available.

Here is a table with bandwidth consumption of various codecs:

Codec Bit Rate Nominal Ethernet Bandwidth (one direction)
G.723.1 5.3 Kbps 20.8 Kbps
G.723.1 6.4 Kbps 21.9 Kbps
G.729 8 Kbps 31.2 Kbps
iLBC 15 Kbps 27.7 Kbps
G.728 16 Kbps 31.5 Kbps
G.726 24 Kbps 47.2 Kbps
G.726 32 Kbps 55.2 Kbps
G.711 ULaw and ALaw 64 Kbps 87.2 Kbps

The GSM codec is similar to iLBC and G.729, but has problems with loosing voice in one direction, so not advisable. Speex is also low bandwidth, but the voice quality varies very much.

G.728 and G.729 are licensed codecs, so not all clients support them.


Here's a small overview on softphones and how to configure them:

Firefly (Windows only, SIP and IAX2 protocol)

Image:Firefly-6.jpg You will see this screen during the installation. Remember to tick "Run Virbiage Soft Phone when Windows starts".
Image:Firefly-9.jpg Once firefly starts, it will ask you, what network you want to use. Chose "Third Party Network".
Image:Firefly-10.jpg Here are the settings you need to fill out:
  • Network Name - not important, just set it to "WanCom"
  • Type - SIP or IAX2, you can choose, if you have trouble getting SIP to work, like no voice when calling, try using IAX2
  • Server Address - wancom.thewan.net
  • Username - the username you supplied, when you asked for the account.
  • Password - the password you were supplied with.

Happy phoning :)

To get G.729 to work with firefly, you can download g729.dll and save it to the directory, where Firefly is installed. Now just restart Firefly and the G.729 option should be clickable.

Grandstream Budgetone IP phone and Handytone Adaptor

SIP Server: wancom.thewan.net
Outbound Proxy: wancom.thewan.net
SIP User ID: Enter your username here.
Authenticate ID: Enter your username here.
Authenticate Password: Enter the password you got assigned here
Name: Your name, optionally

Beyond these fields, it might be important to prioritize low-bandwidth codecs, so iLBC and G.729 first in the order, then G.726-32 and PCMU and PCMA at last. Other codecs are not interesting for us, since we don't have them licensed. Also make sure that the field Send DMTF is set to "via SIP INFO", for functions like HOLD to function properly.

Kiax (Linux only, IAX2 protocol)

Kphone (Linux only, SIP protocol)

  • Download
    • Debian: apt-get install kphone

Snom 320 IP phone

Snom 360 softphone applcation (Windows, SIP protocol)

X-ten X-lite (Linux, Mac, Windows, SIP protocol)

Image:X-lite_1.jpg The first thing you will be presented with after starting X-lite is the Audio Tuning Wizard.
Image:X-lite_2.jpg Select the appropriate soundcard or headset and follow the instructions.
Image:X-lite_3.jpg After finishing the Audio Tuning Wizard, X-lite will ask you for the the SIP proxy details. The fields to fill out here are:
  • Enabled - has to be set to yes
  • Display Name - this is just your name
  • Username - the login, as supplied when you got your account
  • Authorization User - again, your login
  • Password - the supplied password
  • Domain/Realm - X-lite doesn't like, when this isn't filled out, so insert "wancom" here.
  • SIP Proxy - the proxy you are connecting to. This is wancom.thewan.net.
  • Out Bound Proxy - same proxy again. This is wancom.thewan.net.

Happy phoning :)