Luon's VoIP setup

Our VoIP setup is currently based on Jabber/XMPP using ejabberd and SIP using Asterisk.

At the moment the Jabber/XMPP-part is fully federated, but the SIP-part it is not. Hence, it is only possible to call other Luon-users via SIP.

Below we go into more detail of the SIP-setup.

Account

The main account on which you will be reachable is sips request:<username>@luon.net. This account does not exist by default, you have to request it!

If you have multiple devices, you can addionally request an account per device that is coupled to your main account and will be of the form sips:<username>-<devicename>@luon.net. When the main account is called, all registered devices will be called.

Additionally, everyone will have a phone number, useful for SIP softphones that can only dial numbers. The number will correspond to your UID on the Luon servers.

The following account settings need to be filled in to connect:

  • SIP address: <username>@luon.net (or <username>-<devicename>@luon.net)
  • port: 5061
  • transport: (START)TLS
  • ignore TLS errors: yes[1]
  • auto-detect STUN: yes

However, only if you are allowed to provide the full SIP address (i.e. sips:<username>@luon.net), it should be enough to figure the rest out.

1: This is necessary for now, since Asterisk seems to have some issue with chained certificates.

Dialing

To dial someone, just call to <number>/<username> (using the default account) or sips:<number>@luon.net/sips:<username>@luon.net.

Extra services

There are some extra services available:

  • 100/chat: conference room
  • 200/echo: echo service
  • 300/test: sound test service

External dialing

You can try to dial to other SIP accounts, but there is no guarantee that that will work just yet. E.g. you can call to sip:someuser@domain.tld.

Jabber/XMPP (Jingle) to SIP gateway

You can use the Asterisk bot on Jabber/XMPP to dial out from Jabber to SIP. Just add asterisk@luon.net to your roster and call it using Jingle (voice call). The bot will answer and ask for the extension to dial. You can enter a username, extension, or even try an external SIP address and it will forward the call over SIP.